Loudness adjustment for downmixed audio content

ABSTRACT

Audio content coded for a reference speaker configuration is downmixed to downmix audio content coded for a specific speaker configuration. One or more gain adjustments are performed on individual portions of the downmix audio content coded for the specific speaker configuration. Loudness measurements are then performed on the individual portions of the downmix audio content. An audio signal that comprises the audio content coded for the reference speaker configuration and downmix loudness metadata is generated. The downmix loudness metadata is created based at least in part on the loudness measurements on the individual portions of the downmix audio content.

CROSS REFERENCE TO RELATED APPLICATIONS

This application is a divisional of U.S. patent application Ser. No.16/115,292, filed Aug. 28, 2018, which is continuation of U.S. patentapplication Ser. No. 15/625,749, filed Jun. 16, 2017, now U.S. Pat. No.10,070,243, which is continuation of U.S. patent application Ser. No.15/091,366, filed Apr. 5, 2016, now U.S. Pat. No. 9,686,624, which iscontinuation of Ser. No. 14/916,522, filed Mar. 3, 2016, now U.S. Pat.No. 9,521,501, which is U.S. National Stage Entry of InternationalApplication No. PCT/US2014/054718, filed Sep. 9, 2014, which claimspriority to U.S. Provisional Application Nos. 61/938,043, filed Feb. 10,2014, 61/892,313, filed Oct. 17, 2013, 61/891,324, filed Oct. 15, 2013and 61/877,230, filed Sep. 12, 2013, each of which is incorporatedreference in its entirety.

TECHNOLOGY

The present invention pertains generally to processing audio signals andpertains more specifically to techniques that may be used to applyingdynamic range control and other types of audio processing operations toaudio signals in any of a wide variety of playback environments.

BACKGROUND

The increasing popularity of media consumer devices has created newopportunities and challenges for the creators and distributors of mediacontent for playback on those devices, as well as for the designers andmanufacturers of the devices. Many consumer devices are capable ofplaying back a broad range of media content types and formats includingthose often associated with high-quality, wide bandwidth and widedynamic range audio content for HDTV, Blu-ray or DVD. Media processingdevices may be used to play back this type of audio content either ontheir own internal acoustic transducers or on external transducers suchas headphones; however, they generally cannot reproduce this contentwith consistent loudness and intelligibility across varying media formatand content types.

The approaches described in this section are approaches that could bepursued, but not necessarily approaches that have been previouslyconceived or pursued. Therefore, unless otherwise indicated, it shouldnot be assumed that any of the approaches described in this sectionqualify as prior art merely by virtue of their inclusion in thissection. Similarly, issues identified with respect to one or moreapproaches should not assume to have been recognized in any prior art onthe basis of this section, unless otherwise indicated.

BRIEF DESCRIPTION OF DRAWINGS

The present invention is illustrated by way of example, and not by wayof limitation, in the figures of the accompanying drawings and in whichlike reference numerals refer to similar elements and in which:

FIG. 1A and FIG. 1B illustrate an example audio decoder and an exampleaudio encoder, respectively;

FIG. 2A and FIG. 2B illustrate example dynamic range compression curves;

FIG. 3 illustrates example processing logic of determination/calculationof combined DRC and limiting gains;

FIG. 4 illustrates example differential coding of gains;

FIG. 5 illustrates an example codec system comprising an audio encoderand an audio decoder;

FIG. 6A through FIG. 6D illustrate example process flows; and

FIG. 7 illustrates an example hardware platform on which a computer or acomputing device as described herein may be implemented.

DESCRIPTION OF EXAMPLE EMBODIMENTS

Example embodiments, which relate to applying dynamic range control andother types of audio processing operations to audio signals in any of awide variety of playback environments, are described herein. In thefollowing description, for the purposes of explanation, numerousspecific details are set forth in order to provide a thoroughunderstanding of the present invention. It will be apparent, however,that the present invention may be practiced without these specificdetails. In other instances, well-known structures and devices are notdescribed in exhaustive detail, in order to avoid unnecessarilyoccluding, obscuring, or obfuscating the present invention.

Example embodiments are described herein according to the followingoutline:

-   -   1. GENERAL OVERVIEW    -   2. DYNAMIC RANGE CONTROL    -   3. AUDIO DECODER    -   4. AUDIO ENCODER    -   5. DYNAMIC RANGE COMPRESSION CURVES    -   6. DRC GAINS, GAIN LIMITING AND GAIN SMOOTHING    -   7. INPUT SMOOTHING AND GAIN SMOOTHING    -   8. DRC OVER MULTIPLE FREQUENCY BANDS    -   9. VOLUME ADJUSTMENT IN LOUDNESS DOMAIN    -   10. DOWNMIX LOUDNESS ADJUSTMENT    -   11. ADDITIONAL OPERATIONS RELATED TO GAINS    -   12. SPECIFIC AND BROADBAND (OR WIDEBAND) LOUDNESS LEVELS    -   13. INDIVIDUAL GAINS FOR INDIVIDUAL SUBSETS OF CHANNELS    -   14. AUDITORY SCENE ANALYSIS    -   15. LOUDNESS LEVEL TRANSITIONS    -   16. RESET    -   17. ENCODER-PROVIDED GAINS    -   18. EXAMPLE SYSTEM AND PROCESS FLOWS    -   19. IMPLEMENTATION MECHANISMS—HARDWARE OVERVIEW    -   20. EQUIVALENTS, EXTENSIONS, ALTERNATIVES AND MISCELLANEOUS        1. General Overview

This overview presents a basic description of some aspects of anembodiment of the present invention. It should be noted that thisoverview is not an extensive or exhaustive summary of aspects of theembodiment. Moreover, it should be noted that this overview is notintended to be understood as identifying any particularly significantaspects or elements of the embodiment, nor as delineating any scope ofthe embodiment in particular, nor the invention in general. Thisoverview merely presents some concepts that relate to the exampleembodiment in a condensed and simplified format, and should beunderstood as merely a conceptual prelude to a more detailed descriptionof example embodiments that follows below. Note that, although separateembodiments are discussed herein, any combination of embodiments and/orpartial embodiments discussed herein may be combined to form furtherembodiments.

In some approaches, an encoder assumes that audio content is beingencoded for a particular environment for the purpose of dynamic rangecontrol, and determines audio processing parameters such as gains fordynamic range control, etc., for the particular environment. The gainsdetermined by the encoder under these approaches typically have beensmoothed with some time constants (e.g., in an exponential decayfunction, etc.), over some time intervals, etc. In addition, the gainsdetermined by the encoder under these approaches may have beenincorporated for gain limiting which ensure loudness levels to be nomore than the clipping level for the assumed environment. Accordingly,the gains encoded with audio information into an audio signal by theencoder under these approaches are results of many different influencesand irreversible. A decoder receiving the gains under these approacheswould not be able to distinguish which part of the gains are for dynamicrange control, which part of the gains are for gain smoothing, whichpart of the gains are for gain limiting, etc.

Under techniques as described herein, an audio encoder does not assumethat only a specific playback environment at audio decoders needs to besupported. In an embodiment, the audio encoder transmits an encodedaudio signal with audio content from which correct loudness levels(e.g., without clipping, etc.) can be determined. The audio encoder alsotransmits one or more dynamic range compression curves to the audiodecoders. Any of the one or more dynamic range compression curves may bestandard-based, proprietary, customized, content-provider-specific, etc.Reference loudness levels, attack times, release times, etc., may betransmitted by the audio encoder as a part of, or in conjunction with,the one or more dynamic range compression curves. Any of the referenceloudness levels, attack times, release times, etc., may bestandard-based, proprietary, customized, content-provider-specific, etc.

In some embodiments, the audio encoder implements auditory sceneanalysis (ASA) techniques, and uses the ASA techniques to detectauditory events in the audio content, and transmits one or more ASAparameters that describe the detected auditory events to the audiodecoders.

In some embodiments, the audio encoder can also be configured to detectreset events in the audio content, and transmit indications of the resetevents in a time-synchronous manner with the audio content to adownstream device such as an audio decoder, etc.

In some embodiments, the audio encoder can be configured to compute oneor more sets of gains (e.g., DRC gains, etc.) for individual portions(e.g., audio data blocks, audio data frames, etc.) of the audio contentand encode the sets of gains with the individual portions of the audiocontent into the encoded audio signal. In some embodiments, the sets ofgains generated by the audio encoder correspond to one or more differentgain profiles. In some embodiments, Huffman coding, differential coding,etc., may be used to code the sets of gains into, or read the sets ofgains from, components, subdivisions, etc., of audio data frames. Thesecomponents, subdivision, etc., may be referred to as sub-frames in theaudio data frames. Different sets of gains may correspond to differentsets of sub-frames. Each set of gains, or each set of sub-frames, maycomprise two or more temporal components (e.g., sub-frames, etc.). Insome embodiments, a bitstream formatter in an audio encoder as describedherein may write, with one or more for-loops, one or more sets of gainstogether as differential data codes into one or more sets of sub-framesin audio data frames; correspondingly, a bitstream parser in an audiodecoder as described herein may read any of the one or more sets ofgains coded as the differential data codes from the one or more sets ofsub-frames in the audio data frames.

In some embodiments, the audio encoder determines dialogue loudnesslevels in audio content that is to be encoded into the encoded audiosignal, and transmits the dialogue loudness levels with the audiocontent to the audio decoders.

In some embodiments, the audio content is encoded in the encoded audiosignal for a reference speaker configuration (e.g. a surround soundconfiguration, a 5.1 speaker configuration, etc.) that comprises moreaudio channels or speakers than those (e.g., a two channel headsetconfiguration, etc.) with which a large number of audio decoders (e.g.,mobile phones, tablet computers, etc.) operate. Loudness levels asmeasured in the reference speaker configuration for individual portionsof the audio content may be different from loudness levels as measuredin the specific speaker configuration such as a two-channelconfiguration, etc., for the same individual portions of the audiocontent, even if the same gain adjustments are made in both speakerconfigurations.

In some embodiments, an audio encoder as described herein is configuredto provide downmix related metadata (e.g., comprising one or moredownmix loudness parameters, etc.) to downstream audio decoders. Thedownmix related metadata from the audio encoder (150) can be used by adownstream audio decoder to efficiently and consistently perform (e.g.,in real time, in near real time, etc.) additional downmix related gainadjustment operations for the purpose of producing relatively accuratetarget loudness levels in a downmix sound output. The additional downmixrelated gain adjustment operations may be used by the downstream audiodecoder to prevent inconsistencies in measured loudness levels betweenthe reference speaker configuration and the decoder's specific speakerconfiguration.

Techniques as described herein do not require audio decoders to belocked in with (e.g., irreversible, etc.) audio processing which mayhave been performed by an upstream device such as an audio encoder,etc., while assuming a hypothetic playback environment, scenario, etc.,at a hypothetic audio decoder. The decoder as described herein may beconfigured to customize the audio processing operations based on aspecific playback scenario, for example, in order to differentiatedifferent loudness levels existing in audio content, minimize loss ofaudio perceptual quality at or near boundary loudness levels (e.g.,minimum or maximum loudness levels, etc.), maintain spatial balanceamong channels or subsets of channels, etc.

An audio decoder that receives the encoded audio signal with the dynamicrange compression curves, reference loudness levels, attack times,release times, etc., can determine a specific playback environment thatis in use at the decoder, and select a specific compression curve with acorresponding reference loudness level corresponding to the specificplayback environment.

The decoder can compute/determine loudness levels in individual portions(e.g., audio data blocks, audio data frames, etc.) of the audio contentextracted from the encoded audio signal, or obtain the loudness levelsin the individual portions of the audio content if the audio encoder hascomputed and provided the loudness levels in the encoded audio signal.Based on one or more of the loudness levels in the individual portionsof the audio content, loudness levels in previous portions of the audiocontent, loudness levels in subsequent portions of the audio content ifavailable, the specific compression curve, a specific profile related tothe specific playback environment or scenario, etc., the decoderdetermines audio processing parameters such as gains for dynamic rangecontrol (or DRC gains), attack times, release times, etc. The audioprocessing parameters also can include adjustments for aligning dialogueloudness levels to a specific reference loudness level (which may beuser adjustable) for the specific playback environment.

The decoder applies audio processing operations including (e.g.,multi-channel, multi-band, etc.) dynamic range control, dialogue leveladjustments, etc., with the audio processing parameters. The audioprocessing operations performed by the decoder may further include butare not limited to only: gain smoothing based on the attack and releasetimes provided as a part of, or in conjunction with, the selecteddynamic range compression curve, gain limiting for preventing clipping,etc. Different audio processing operations may be performed withdifferent (e.g., adjustable, threshold-dependent, controllable, etc.)time constants. For example, gain limiting for preventing clipping maybe applied to individual audio data blocks, individual audio dataframes, etc., with relatively short time constants (e.g., instantaneous,approximately 5.3 milliseconds, etc.).

In some embodiments, the decoder can be configured to extract ASAparameters (e.g., temporal locations of auditory event boundaries,time-dependent values of an event certainty measure, etc.) from metadatain an encoded audio signal, and control the speed of gain smoothing inauditory events based on the extracted ASA parameters (e.g., use shorttime constants for attacks at auditory events boundaries, use long timeconstants to slow down gain smoothing within an auditory event, etc.).

In some embodiments, the decoder also maintains a histogram ofinstantaneous loudness levels for a certain time interval or window, anduses the histogram to control the speed of gain changes in loudnesslevel transitions between programs, between a program and a commercial,etc., for example, by modifying the time constants.

In some embodiments, the decoder supports more than one speakerconfiguration (e.g., portable mode with speakers, portable mode withheadphones, stereo mode, multi-channel mode, etc.). The decoder may beconfigured to maintain the same loudness levels between two differentspeaker configurations (e.g., between a stereo mode and a multi-channelmode, etc.), for example, when playing back the same audio content. Theaudio decoder may use one or more downmix equations to downmixmulti-channel audio content, as received from an encoded audio signalfor a reference speaker configuration for which the multi-channel audiocontent was coded to a specific speaker configuration at the audiodecoder.

In some embodiments, automatic gain control (AGC) may be disabled in anaudio decoder as described herein.

In some embodiments, mechanisms as described herein form a part of amedia processing system, including but not limited to: an audiovisualdevice, a flat panel TV, a handheld device, game machine, television,home theater system, tablet, mobile device, laptop computer, netbookcomputer, cellular radiotelephone, electronic book reader, point of saleterminal, desktop computer, computer workstation, computer kiosk,various other kinds of terminals and media processing units, etc.

Various modifications to the preferred embodiments and the genericprinciples and features described herein will be readily apparent tothose skilled in the art. Thus, the disclosure is not intended to belimited to the embodiments shown, but is to be accorded the widest scopeconsistent with the principles and features described herein.

2. Dynamic Range Control

Without customized dynamic range control, input audio information (e.g.,PCM samples, time-frequency samples in a QMF matrix, etc.) is oftenreproduced at a playback device at loudness levels that areinappropriate for the playback device's specific playback environment(that is, including the device's physical and/or mechanical playbacklimitations), as the playback device's specific playback environmentmight be different from a target playback environment for which theencoded audio content had been coded at an encoding device.

Techniques as described herein can be used to support dynamic rangecontrol of a wide variety of audio content customized to any of a widevariety of playback environments while maintaining perceptual qualitiesof the audio content.

Dynamic Range Control (DRC) refers to time-dependent audio processingoperations that alter (e.g., compress, cut, expand, boost, etc.) aninput dynamic range of loudness levels in audio content into an outputdynamic range that is different from the input dynamic range. Forexample, in a dynamic range control scenario, soft sounds may be mapped(e.g., boosted, etc.) to higher loudness levels and loud sounds may bemapped (e.g., cut, etc.) to lower loudness values. As a result, in aloudness domain, an output range of loudness levels becomes smaller thanthe input range of loudness levels in this example. In some embodiments,the dynamic range control, however, may be reversible so that theoriginal range is restored. For example, an expansion operation may beperformed to recover the original range so long as mapped loudnesslevels in the output dynamic range, as mapped from original loudnesslevels, are at or below a clipping level, each unique original loudnesslevel is mapped to a unique output loudness level, etc.

DRC techniques as described herein can be used to provide a betterlistening experience in certain playback environments or situations. Forexample, soft sounds in a noisy environment may be masked by the noisethat renders the soft sounds inaudible. Conversely, loud sounds may notbe desired in some situations, for example, bothering neighbors. Manydevices, typically with small form-factor loudspeakers, cannot reproducesound at high output levels. In some cases, the lower signal levels maybe reproduced below the human hearing threshold. The DRC techniques mayperform mapping of input loudness levels to output loudness levels basedon DRC gains (e.g., scaling factors that scale audio amplitudes, boostratios, cut ratios, etc.) looked up with a dynamic range compressioncurve.

A dynamic range compression curve refers to a function (e.g., a lookuptable, a curve, a multi-segment piecewise lines, etc.) that mapsindividual input loudness levels (e.g., of sounds other than dialogues,etc.) as determined from individual audio data frames to individualgains or gains for dynamic range control. Each of the individual gainsindicates an amount of gain to be applied to a corresponding individualinput loudness level. Output loudness levels after applying theindividual gains represent target loudness levels for audio content inthe individual audio data frames in a specific playback environment.

In addition to specifying mappings between gains and loudness levels, adynamic range compression curve may include, or may be provided with,specific release times and attack times in applying specific gains. Anattack refers to an increase of signal energy (or loudness) betweensuccessive time samples, whereas a release refers to a decrease ofenergy (or loudness) between successive time samples. An attack time(e.g., 10 milliseconds, 20 milliseconds, etc.) refers to a time constantused in smoothing DRC gains when the corresponding signal is in attackmode. A release time (e.g., 80 milliseconds, 100 milliseconds, etc.)refers to a time constant used in smoothing DRC gains when thecorresponding signal is in release mode. In some embodiments,additionally, optionally or alternatively, the time constants are usedfor smoothing of the signal energy (or loudness) prior to determiningthe DRC gain.

Different dynamic range compression curves may correspond to differentplayback environments. For example, a dynamic range compression curvefor a playback environment of a flat panel TV may be different from adynamic range compression curve for a playback environment of a portabledevice. In some embodiments, a playback device may have two or moreplayback environments. For example, a first dynamic range compressioncurve for a first playback environment of a portable device withspeakers may be different from a second dynamic range compression curvefor a second playback environment of the same portable device withheadset.

3. Audio Decoder

FIG. 1A illustrates an example audio decoder 100 comprising a dataextractor 104, dynamic range controller 106, an audio renderer 108, etc.

In some embodiments, the data extractor (104) is configured to receivean encoded input signal 102. An encoded input signal as described hereinmay be a bit stream that contains encoded (e.g., compressed, etc.) inputaudio data frames and metadata. The data extractor (104) is configuredto extract/decode input audio data frames and metadata from the encodedinput signal (102). Each of the input audio data frames comprises aplurality of coded audio data blocks each of which represents aplurality of audio samples. Each frame represents a (e.g., constant)time interval comprising a certain number of audio samples. The framesize may vary with the sample rate and coded data rate. The audiosamples are quantized audio data elements (e.g., input PCM samples,input time-frequency samples in a QMF matrix, etc.) representingspectral content in one, two or more (audio) frequency bands orfrequency ranges. The quantized audio data elements in the input audiodata frames may represent pressure waves in a digital (quantized)domain. The quantized audio data elements may cover a finite range ofloudness levels at or below a largest possible value (e.g., a clippinglevel, a maximum loudness level, etc.).

The metadata can be used by a wide variety of recipient decoder toprocess the input audio data frames. The metadata may include a varietyof operational parameters relating to one or more operations to beperformed by the decoder (100), one or more dynamic range compressioncurves, normalization parameters relating to dialogue loudness levelsrepresented in the input audio data frames, etc. A dialogue loudnesslevel may refer to a (e.g., psychoacoustic, perceptual, etc.) level ofdialogue loudness, program loudness, average dialogue loudness, etc., inan entire program (e.g., a movie, a TV program, a radio broadcast,etc.), a portion of a program, a dialogue of a program, etc.

The operation and functions of the decoder (100), or some or all of themodules (e.g., the data extractor 104, the dynamic range controller 106,etc.), may be adapted in response to the metadata extracted from theencoded input signal (102). For example, the metadata—including but notlimited to dynamic range compression curves, dialogue loudness levels,etc.—may be used by the decoder (100) to generate output audio dataelements (e.g., output PCM samples, output time-frequency samples in aQMF matrix, etc.) in the digital domain. The output data elements canthen be used to drive audio channels or speakers to achieve a specifiedloudness or reference reproduction level during playback in a specificplayback environment.

In some embodiments, the dynamic range controller (106) is configured toreceive some or all of the audio data elements in the input audio dataframes and the metadata, perform audio processing operations (e.g.,dynamic range control operations, gain smoothing operations, gainlimiting operations, etc.) on the audio data elements in the input audiodata frames based at least in part on the metadata extracted from theencoded audio signal (102), etc.

In some embodiments, the dynamic range controller (106) may comprise aselector 110, a loudness calculator 112, DRC gain unit 114, etc. Theselector (110) may be configured to determine a speaker configuration(e.g., flat panel mode, portable device with speakers, portable devicewith headphones, a 5.1 speaker configuration, a 7.1 speakerconfiguration, etc.) relating to a specific playback environment at thedecoder (100), select a specific dynamic range compression curve fromthe dynamic range compression curves extracted from the encoded inputsignal (102), etc.

The loudness calculator (112) may be configured to calculate one or moretypes of loudness levels as represented by the audio data elements inthe input audio data frames. Examples of types of loudness levelsinclude, but are not limited only to: any of individual loudness levelsover individual frequency bands in individual channels over individualtime intervals, broadband (or wideband) loudness levels over a broad (orwide) frequency range in individual channels, loudness levels asdetermined from or smoothed over an audio data block or frame, loudnesslevels as determined from or smoothed over more than one audio datablock or frame, loudness levels smoothed over one or more timeintervals, etc. Zero, one or more of these loudness levels may bealtered for the purpose of dynamic range control by the decoder (100).

To determine the loudness levels, the loudness calculator (112) candetermine one or more time-dependent physical sound wave properties suchas spatial pressure levels at specific audio frequencies, etc., asrepresented by the audio data elements in the input audio data frames.The loudness calculator (112) can use the one or more time-varyingphysical wave properties to derive one or more types of loudness levelsbased on one or more psychoacoustic functions modeling human loudnessperception. A psychoacoustic function may be a non-linear function—asconstructed based on a model of the human auditory system—thatconverts/maps specific spatial pressure levels at specific audiofrequencies to specific loudness for the specific audio frequencies,etc.

A (e.g., broadband, wideband, etc.) loudness level over multiple (audio)frequencies or multiple frequency bands may be derived throughintegration of specific loudness level over the multiple (audio)frequencies or multiple frequency bands. Time-averaged, smoothed, etc.,loudness levels over one or more time intervals (e.g., longer than thatrepresented by audio data elements in an audio data block or frame,etc.) may be obtained by using one or more smoothing filters that areimplemented as a part of the audio processing operations in the decoder(100).

In an example embodiment, specific loudness levels for differentfrequency bands may be calculated per audio data block of certain (e.g.,256, etc.) samples. Pre-filters may be used to apply frequency weighting(e.g., similar to IEC B-weighting, etc.) to the specific loudness levelsin integrating the specific loudness levels into a broadband (orwideband) loudness level. A summation of broad loudness levels over twoor more channels (e.g., left front, right front, center, left surround,right surround, etc.) may be performed to provide an overall loudnesslevel of the two or more channels.

In some embodiments, an overall loudness level may refer to a broadband(wideband) loudness level in a single channel (e.g., center, etc.) of aspeaker configuration. In some embodiments, an overall loudness levelmay refer to a broadband (or wideband) loudness level in a plurality ofchannels. The plurality of channels may be all channels in a speakerconfiguration. Additionally, optionally or alternatively, the pluralityof channels may comprise a subset of channels (e.g., a subset ofchannels comprising left front, right front, and low frequency effect(LFE); a subset of channels comprising left surround and right surround;etc.) in a speaker configuration.

A (e.g., broadband, wideband, overall, specific, etc.) loudness levelmay be used as input to look up a corresponding (e.g., static,pre-smoothing, pre-limiting, etc.) DRC gain from the selected dynamicrange compression curve. The loudness level to be used as input to lookup the DRC gain may be first adjusted or normalized with respect to adialogue loudness level from the metadata extracted from the encodedaudio signal (102).

In some embodiments, the DRC gain unit (114) may be configured with aDRC algorithm to generate gains (e.g., for dynamic range control, forgain limiting, for gain smoothing, etc.), apply the gains to one or moreloudness levels in the one or more types of loudness levels representedby the audio data elements in the input audio data frames to achievetarget loudness levels for the specific playback environment, etc. Theapplication of gains as described herein (e.g., DRC gains, etc.) may,but is not required to, happen in the loudness domain. In someembodiments, gains may be generated based on the loudness calculation(which may be in Sone or just the SPL value compensated for the dialogloudness level, for example, with no conversion), smoothed and applieddirectly to the input signal. In some embodiments, techniques asdescribed herein may apply the gains to a signal in the loudness domain,and then convert the signal from the loudness domain back to the(linear) SPL domain and calculate corresponding gains that are to beapplied to the signal by assessing the signal before and after the gainwas applied to the signal in the loudness domain. The ratio (ordifference when represented in a logarithmic dB representation) thendetermines the corresponding gain for the signal.

In some embodiments, the DRC algorithm operates with a plurality of DRCparameters. The DRC parameters include the dialogue loudness level thathas already been computed and embedded into the encoded audio signal(102) by an upstream encoder (e.g., 150, etc.) and can be obtained fromthe metadata in the encoded audio signal (102) by the decoder (100). Thedialogue loudness level from the upstream encoder indicates an averagedialogue loudness level (e.g., per program, relative to the energy of afull-scale 1 kHz sine wave, relative to the energy of a referencerectangular wave, etc.). In some embodiments, the dialogue loudnesslevel extracted from the encoded audio signal (102) may be used toreduce inter-program loudness level differences. In an embodiment, thereference dialogue loudness level may be set to the same value betweendifferent programs in the same specific playback environment at thedecoder (100). Based on the dialogue loudness level from the metadata,the DRC gain unit (114) can apply a dialogue loudness related gain toeach audio data block in a program such that an output dialogue loudnesslevel averaged over a plurality of audio data blocks of the program israised/lowered to a (e.g., pre-configured, system default,user-configurable, profile dependent, etc.) reference dialogue loudnesslevel for the program.

In some embodiments, the DRC gains may be used to address intra-programloudness level differences by boosting or cutting input loudness levelsin soft and/or loud sounds in accordance with the selected dynamic rangecompression curve. One or more of these DRC gains may becomputed/determined by the DRC algorithm based on the selected dynamicrange compression curve and (e.g., broadband, wideband, overall,specific, etc.) loudness levels as determined from one or more of thecorresponding audio data blocks, audio data frames, etc.

Loudness levels used to determine (e.g., static, pre-smoothing, pre-gainlimiting, etc.) DRC gains by looking up the selected dynamic rangecompression curve may be calculated on short intervals (e.g.,approximately 5.3 milliseconds, etc.). The integration time of the humanauditory system (e.g., approximately 200 milliseconds, etc.) may be muchlonger. The DRC gains obtained from the selected dynamic rangecompression curve may be smoothed with a time constant to take intoaccount the long integration time of the human auditory system. Toeffectuate fast rates of changes (increases or decreases) in loudnesslevels, short time constants may be used to cause changes in loudnesslevels in short time intervals corresponding to the short timeconstants. Conversely, to effectuate slow rates of changes (increases ordecreases) in loudness levels, long time constants may be used tochanges in loudness levels in long time intervals corresponding to thelong time constants.

The human auditory system may react to increasing loudness levels anddecreasing loudness levels with different integration time. In someembodiments, different time constants may be used for smoothing thestatic DRC gains looked up from the selected dynamic range compressioncurves, depending on whether the loudness level will be increasing ordecreasing. For example, in correspondence with the characteristics ofthe human auditory system, attacks (loudness level increasing) aresmoothed with relatively short time constants (e.g., attack times,etc.), whereas releases (loudness level decreasing) are smoothed withrelatively long time constants (e.g., release time, etc.).

A DRC gain for a portion (e.g., one or more of audio data blocks, audiodata frames, etc.) of audio content may be calculated using a loudnesslevel determined from the portion of audio content. The loudness levelto be used for looking up in the selected dynamic range compressioncurve may be first adjusted with respect to (e.g., in relation to, etc.)a dialogue loudness level (e.g., in a program of which the audio contentis a part, etc.) in the metadata extracted from the encoded audio signal(102).

A reference dialogue loudness level (e.g., −31 dB_(FS) in the “Line”mode, −20 dB_(FS) in the “RF” mode, etc.) may be specified orestablished for the specific playback environment at the decoder (100).Additionally, alternatively or optionally, in some embodiments, usersmay be given control over setting or changing the reference dialogueloudness level at the decoder (100).

The DRC gain unit (114) can be configured to determine a dialogueloudness related gain to the audio content to cause a change from theinput dialogue loudness level to the reference dialogue loudness levelas the output dialogue loudness level.

In some embodiments, the DRC gain unit (114) may be configured to handlepeak levels in the specific playback environment at the decoder (100)and adjusts the DRC gains to prevent clipping. In some embodiments,under a first approach, if the audio content extracted from the encodedaudio signal (102) comprise audio data elements for a referencemulti-channel configuration with more channels than those of thespecific speaker configuration at the decoder (100), downmixing from thereference multi-channel configuration to the specific speakerconfiguration may be performed before determining and handle peak levelsfor the purpose of clipping prevention. Additionally, optionally, oralternatively, in some embodiments, under a second approach, if theaudio content extracted from the encoded audio signal (102) compriseaudio data elements for a reference multi-channel configuration withmore channels than those of the specific speaker configuration at thedecoder (100), downmix equations (e.g., ITU stereo downmix,matrixed-surround compatible downmix, etc.) may be used to obtain thepeak levels for the specific speaker configuration at the decoder (100).The peak level may be adjusted to reflect the change from the inputdialogue loudness level to the reference dialogue loudness level as theoutput dialogue loudness level. A maximum allowed gain without causingclipping (e.g., for an audio data block, for an audio data frame, etc.)may be determined based at least in part on an inverse (e.g., multipliedwith −1, etc.) of the peak level. Thus, an audio decoder undertechniques as described herein can be configured to determine peaklevels accurately and apply clipping prevention specifically for theplayback configuration at the decoder side; neither the audio decodernor the audio encoder is required to make hypothetical assumptions aboutany worst-case scenarios at hypothetical decoders. In particular, thedecoder in the first approach as described above can determine peaklevels accurately and apply clip prevention after downmixing withoutusing downmixing equations, downmix channel gains, etc., which would beused under the second approach as described above,

In some embodiments, a combination of the adjustments to the dialogueloudness level and the DRC gain prevents clipping in the peak level,possibly even in the worst-case downmix (e.g., producing the largestpeak levels after downmixing, producing the largest downmix channelgains, etc.). However, in some other embodiments, the combination of theadjustments to the dialogue loudness level and the DRC gain may stillnot be sufficient in preventing clipping in the peak level. In theseembodiments, the DRC gain may be replaced (e.g., capped, etc.) by thehighest gain that does prevent clipping in the peak level.

In some embodiments, the DRC gain unit (114) is configured to get timeconstants (e.g., attack times, release times, etc.) from the metadataextracted from the encoded audio signal (102). The DRC gain, the timeconstants, the maximum allowed gain, etc., may be used by the DRC gainunit (114) to perform DRC, gain smoothing, gain limiting, etc.

For example, the application of the DRC gain may be smoothed with afilter controlled by a time constant. A gain limiting operation may beimplemented by a min( ) function that takes the lower of a gain to beapplied and a maximum allowed gain for the gain, through which the(e.g., pre-limiting, DRC, etc.) gain may be replaced immediately, over arelatively short time interval, etc., with the maximum allowed gain,thereby preventing clipping.

In some embodiments, the audio renderer (108) is configured to generate(e.g., multi-channel, etc.) channel-specific audio data (116) for thespecific speaker configuration after applying gains as determined basedon DRC, gain limiting, gain smoothing, etc., to the input audio dataextracted from the encoded audio signal (102). The channel-specificaudio data (118) may be used to drive speakers, headphones, etc.,represented in the speaker configuration.

Additionally and/or optionally, in some embodiments, the decoder (100)can be configured to perform one or more other operations relating topreprocessing, post-processing, rendering, etc., relating to the inputaudio data.

Techniques as described herein can be used with a variety of speakerconfigurations corresponding to a variety of different surround soundconfigurations (e.g., 2.0, 3.0, 4.0, 4.1, 4.1, 5.1, 6.1, 7.1, 7.2, 10.2,a 10-60 speaker configuration, a 60+ speaker configuration, etc.) and avariety of different rendering environment configurations (e.g., cinema,park, opera houses, concert halls, bars, homes, auditoriums, etc.).

4. Audio Encoder

FIG. 1B illustrates an example encoder 150. The encoder (150) maycomprise an audio content interface 152, a dialogue loudness analyzer154, a DRC reference repository 156, an audio signal encoder 158, etc.The encoder 150 may be a part of a broadcast system, an internet-basedcontent server, an over-the-air network operator system, a movieproduction system, etc.

In some embodiments, the audio content interface (152) is configured toreceive audio content 160, audio content control input 162, etc.,generate an encoded audio signal (e.g., 102) based at least on some orall of the audio content (160), the audio content control input (162),etc. For example, the audio content interface (152) may be used toreceive the audio content (160), the audio content control input (162)from a content creator, a content provider, etc.

The audio content (160) may constitute some or all of overall media datathat comprises audio only, audiovisual, etc. The audio content (160) maycomprise one or more of portions of a program, a program, severalprograms, one or more commercials, etc.

In some embodiments, the dialogue loudness analyzer (154) is configuredto determine/establish one or more dialogue loudness levels of one ormore portions (e.g., one or more programs, one or more commercials,etc.) of the audio content (152). In some embodiments, the audio contentis represented by one or more sets of audio tracks. In some embodiments,dialogue audio content of the audio content is in separate audio tracks.In some embodiments, at least a portion of dialogue audio content of theaudio content is in audio tracks comprising non-dialogue audio content.

The audio content control input (162) may comprise some or all of usercontrol input, control input provided by a system/device external to theencoder (150), control input from a content creator, control input froma content provider, etc. For example, a user such as a mixing engineer,etc., can provide/specify one or more dynamic range compression curveidentifiers; the identifiers may be used to retrieve one or more dynamicrange compression curves that fit the audio content (160) best from adata repository such as a DRC reference repository (156), etc.

In some embodiments, the DRC reference repository (156) is configured tostore DRC reference parameter sets, etc. The DRC reference parametersets may include definition data for one or more dynamic rangecompression curves, etc. In some embodiments, the encoder (150) may(e.g., concurrently, etc.) encode more than one dynamic rangecompression curve into the encoded audio signal (102). Zero, one, ormore of the dynamic range compression curves may be standard-based,proprietary, customized, decoder-modifiable, etc. In an exampleembodiment, both dynamic range compression curves of FIG. 2A and FIG. 2Bcan be (e.g., concurrently, etc.) encoded into the encoded audio signal(102).

In some embodiments, the audio signal encoder (158) can be configured toreceive the audio content from the audio content interface (152), thedialogue loudness levels from the dialogue loudness analyzer (154),etc., retrieve one or more DRC reference parameter sets from the DRCreference repository (156), format audio content into audio datablocks/frames, format the dialogue loudness levels, the DRC referenceparameter sets, etc., into metadata (e.g., metadata containers, metadatafields, metadata structures, etc.), encode the audio data blocks/framesand the metadata into the encoded audio signal (102), etc.

Audio content to be encoded into an encoded audio signal as describedherein may be received in one or more of a variety of source audioformats in one or more of a variety of ways, such as wirelessly, via awired connection, through a file, via an internet download, etc.

An encoded audio signal as described herein can be a part of an overallmedia data bitstream (e.g., for an audio broadcast, an audio program, anaudiovisual program, an audiovisual broadcast, etc.). The media databitstream can be accessed from a server, a computer, a media storagedevice, a media database, a media file, etc. The media data bit streammay be broadcasted, transmitted or received through one or more wirelessor wired network links. A media data bitstream may also be communicatedthrough an intermediary such as one or more of network connections, USBconnections, wide area networks, local area networks, wirelessconnections, optical connections, buses, crossbar connections, serialconnections, etc.

Any of the components depicted (e.g., FIG. 1A, FIG. 1B, etc.) may beimplemented as one or more processes and/or one or more IC circuits(e.g., ASICs, FPGAs, etc.), in hardware, software, or a combination ofhardware and software.

5. Dynamic Range Compression Curves

FIG. 2A and FIG. 2B illustrate example dynamic range compression curvesthat can be used by the DRC gain unit (104) in the decoder (100) toderive the DRC gains from input loudness levels. As illustrated, adynamic range compression curve may be centered around a referenceloudness level in a program in order to provide overall gains that areappropriate for the specific playback environment. Example definitiondata (e.g., in the metadata of the encoded audio signal 102, etc.) ofthe dynamic range compression curve (e.g., including but not limitedonly to any of: boost ratios, cut ratios, attack times, release times,etc.) are shown in the following table, where each profile in aplurality of profiles (e.g., film standard, film light, music standard,music light, speech, etc.) represents a specific playback environment(e.g., at the decoder 100, etc.):

TABLE 1 Profile Film Film Music Music Parameter standard light Standardlight Speech Time Constant Selection Attack Threshold (dB) 15 15 15 1510 Release Threshold (dB) 20 20 20 20 10 Fast Attack Time Constant 10 1010 10 10 (ms) Slow Attack Time Constant 100 100 100 100 100 (ms) SlowRelease Time Constant 3000 3000 10000 3000 1000 (ms) Fast Release TimeConstant 1000 1000 1000 1000 200 (ms) Holdoff period (ms) 53 53 53 53 53Compression Curve Maximum Boost (dB) 6 6 12 12 15 Maximum Boost Range−43 −53 −55 −65 −50 (<=dB) Boost Ratio 2:1 2:1 2:1 2:1 19:15 Boost Range(dB) [−43, . . . , −31] [−53, . . . , −41] [−55, . . . , −31] [−65, . .. , −41] [−50, . . . , −31] Null Band Width (dB) 5 20 5 20 5 Null BandRange (dB) [−31, . . . , −26] [−41, . . . , −21] [−31, . . . , −26][−41, . . . , −21] [−31, . . . , −26] Cut Ratio 2:1 2:1 2:1 2:1 2:1 CutRatio Range (dB) [−26, . . . , −16] [−21, . . . , −11] [−26, . . . ,−16] [−21, . . . , 27] [−26, . . . , −16] Secondary Cut Ratio 20:1920:19 20:19 20:19 Secondary Cut Ratio Range [−16, . . . , 4] [−11, . . ., 9] [−16, . . . , 4] [−16, . . . , 4] (dB) Maximum Cut (dB) −24 −24 −24−24 −24 Maximum Cut Range (>=dB) 4 9 4 27 4

Some embodiments may receive one or more compression curves described interms of loudness levels in dB_(SPL) or dB_(FS) and gains in dB relatingto dB_(SPL), whereas DRC gain calculation is performed in a differentloudness representation that has a non-linear relation with dB_(SPL)loudness levels (e.g. Sone). The compression curve used in the DRC gaincalculation may then be converted to be described in terms of thedifferent loudness representation (e.g. Sone).

6. DRC Gains, Gain Limiting and Gain Smoothing

FIG. 3 illustrates example processing logic of determination/calculationof combined DRC and limiting gains. The processing logic may beimplemented by the decoder (100), the encoder (150), etc. For thepurpose of illustration only, a DRC gain unit (e.g., 114) in a decoder(e.g., 100, etc.) may be used to implement the processing logic.

A DRC gain for a portion (e.g., one or more of audio data blocks, audiodata frames, etc.) of audio content may be calculated using a loudnesslevel determined from the portion of audio content. The loudness levelmay be first adjusted with respect to (e.g., in relation to, etc.) adialogue loudness level (e.g., in a program of which the audio contentis a part, etc.) in the metadata extracted from the encoded audio signal(102). In an example as illustrated in FIG. 3, a difference between theloudness level of the portion of audio content and the dialogue loudnesslevel (“dialnorm”) may be used as an input to look up the DRC gain fromthe selected dynamic range compression curve.

In order to prevent clipping in the output audio data elements in thespecific playback environment, the DRC gain unit (114) may be configuredto handle peak levels in a specific playback scenario (e.g., specific tothe particular combination of the encoded audio signal 102 and theplayback environment at the decoder 100, etc.), which may be one in avariety of possible playback scenarios (e.g., multi-channel scenarios,downmix scenarios, etc.).

In some embodiments, individual peak levels for individual portions ofthe audio content at a particular time resolution (e.g., audio datablock, several audio data blocks, an audio data frame, etc.) may beprovided as a part of the metadata extracted from the encoded audiosignal (102).

In some embodiments, the DRC gain unit (114) can be configured todetermine the peak level in these scenarios and adjusts the DRC gains ifnecessary. During the calculation of the DRC gain, a parallel processmay be used by the DRC gain unit (114) to determine the peak level ofthe audio content. For example, the audio content may be encoded for areference multi-channel configuration that has more channels than thoseof a specific speaker configuration used by the decoder (100). The audiocontent for the more channels of the reference multi-channelconfiguration may be converted into a downmixed audio data (e.g., ITUstereo downmix, matrixed-surround compatible downmix, etc.) to drive thefewer channels for the specific speaker configuration at the decoder(100). In some embodiments, under a first approach, downmixing from thereference multi-channel configuration to the specific speakerconfiguration may be performed before determining and handle peak levelsfor the purpose of clipping prevention. Additionally, optionally, oralternatively, in some embodiments, under a second approach, downmixchannel gains relating to downmixing the audio content may be used as apart of input to adjust, derive, compute, etc., the peak level for thespecific speaker configuration. In an example embodiment, the downmixchannel gains may be derived based at least in part on one or moredownmix equations used to carry out the downmix operation from thereference multi-channel configuration to the specific speakerconfiguration in the playback environment at the decoder (100).

In some media applications, a reference dialogue loudness level (e.g.,−31 dB_(FS) in the “Line” mode, −20 dB_(FS) in the “RF” mode, etc.) maybe specified or assumed for the specific playback environment at thedecoder (100). In some embodiments, users may be given control oversetting or changing the reference dialogue loudness level at the decoder(100).

A dialogue loudness related gain may be applied to the audio content toadjust the (e.g., output) dialogue loudness level to the referencedialogue loudness level. The peak level should be adjusted accordinglyto reflect this adjustment. In an example, the (input) dialogue loudnesslevel may be −23 dB_(FS). In a “Line” mode with a reference dialogueloudness level of −31 dB_(FS), the adjustment to the (input) dialogueloudness level is −8 dB to produce an output dialogue loudness level atthe reference dialogue loudness level. In the “Line” mode, theadjustment to the peak level is also −8 dB, the same as the adjustmentto the dialogue loudness level. In an “RF” mode with a referencedialogue loudness level of −20 dB_(FS), the adjustment to the (input)dialogue loudness level is 3 dB to produce an output dialogue loudnesslevel at the reference dialogue loudness level. In the “RF” mode, theadjustment to the peak level is also 3 dB, the same as the adjustment tothe dialogue loudness level.

A sum of the peak level and a difference between the reference dialogueloudness level (denoted as “dialref”) and the dialogue loudness level(“dialnorm”) in the metadata from the encoded audio signal (102) may beused as an input to compute a maximum (e.g., allowed, etc.) gain for theDRC gain. Since the adjusted peak level is expressed in dB_(FS)(relative to the clipping level at 0 dB_(FS)), the maximum allowed gainwithout causing clipping (e.g., for the current audio data block, forthe current audio data frame, etc.) is simply the inverse (e.g.,multiplied with −1, etc.) of the adjusted peak level.

In some embodiments, the peak level may exceed a clipping level (denotedas 0 dB_(FS)), even if the dynamic range compression curve from whichthe DRC gain was derived is designed to cut loud sounds to a certainextent. In some embodiments, a combination of the adjustments to thedialogue loudness level and the DRC gain prevents clipping in the peaklevel, possibly even in the worst-case downmix (e.g., producing thelargest downmix channel gains, etc.). However, in some otherembodiments, the combination of the adjustments to the dialogue loudnesslevel and the DRC gain may still not be sufficient in preventingclipping in the peak level. In these embodiments, the DRC gain may bereplaced (e.g., capped, etc.) by the highest gain that does preventclipping in the peak level.

In some embodiments, the DRC gain unit (114) is configured to get timeconstants (e.g., attack times, release times, etc.) from the metadataextracted from the encoded audio signal (102). These time constants mayor may not vary with one or more of the dialogue loudness level or thecurrent loudness level of the audio content. The DRC gain looked up fromthe dynamic range compression curve, the time constants, and the maximumgain may be used to perform gain smoothing and limiting operations.

In some embodiments, the DRC gain which may possibly be gain limiteddoes not exceed the maximum peak loudness level in the specific playbackenvironment. The static DRC gain derived from the loudness level may besmoothed with a filter controlled by a time constant. The limitingoperations may be implemented by one or more min( ) functions, throughwhich the (pre-limiting) DRC gain may be replaced immediately, over ashort time interval, etc., with the maximum allowed gain, therebypreventing clipping. The DRC algorithm may be configured to smoothlyrelease from the clipping gain to a lower gain as the peak levels ofincoming audio content moves from above the clipping level to below theclipping level.

One or more different (e.g., real time, two-pass, etc.) implementationsmay be used to carry out the determination/calculation/application ofDRC gains as illustrated in FIG. 3. For the purpose of illustrationonly, the adjustments to the dialogue loudness level, the (e.g., static,etc.) DRC gains, the time-dependent gain variations due to smoothing,gain clipping due to limiting, etc., have been described as combinedgains from the DRC algorithm as described above. However, otherapproaches of applying gains to audio content for controlling dialogueloudness levels (e.g., between different programs, etc.), for dynamicrange control (e.g., for different portions of the same program, etc.),for preventing clipping, for gain smoothing, etc., may be used invarious embodiments. For example, some or all of the adjustments to thedialogue loudness level, the (e.g., static, etc.) DRC gains, thetime-dependent gain variations due to smoothing, gain clipping due tolimiting, etc., may be partially/individually applied, applied inseries, applied in parallel, applied in part series in part parallel,etc.

7. Input Smoothing and Gain Smoothing

In addition to DRC gain smoothing, other smoothing processes undertechniques as described herein may be implemented in variousembodiments. In an example, input smoothing may be used to smooth inputaudio data extracted from the encoded audio signal (102), for examplewith a simple single pole smoothing filter, to obtain a spectrum ofspecific loudness levels that has better temporal characteristics (e.g.,more smooth in time, less spiky in time, etc.) than a spectrum ofspecific loudness levels without input smoothing.

In some embodiments, different smoothing processes as described hereincan use different time constants (e.g., 1 second, 4 seconds, etc.). Insome embodiments, two or more smoothing processes can use a same timeconstant. In some embodiments, time constants used in smoothingprocesses as described herein can be frequency-dependent. In someembodiments, time constants used in smoothing processes as describedherein can be frequency-independent.

One or more smoothing processes may be connected to a reset process thatsupports an automatic or manual reset of the one or more smoothingprocesses. In some embodiments, when a reset occurs in the resetprocess, a smoothing process may speed up smoothing operations byswitching or transferring to a smaller time constant. In someembodiments, when a reset occurs in the reset process, the memory of asmoothing process may be reset to a certain value. This value may be thelast input sample to the smoothing process.

8. DRC Over Multiple Frequency Bands

In some embodiments, specific loudness levels in specific frequencybands can be used to derive corresponding DRC gains in the specificfrequency bands. This, however, may result in timbre changes as thespecific loudness levels can vary significantly in different bands andthus incur different DRC gains, even as a broadband (or wideband)loudness level over all the frequency bands remains constant.

In some embodiments, rather than applying DRC gains that vary withindividual frequency bands, DRC gains that do not vary with frequencybands but vary with time are applied instead. The same time-varying DRCgains are applied across all of the frequency bands. The time-averagedDRC gains of the time-varying DRC gains may be set to the same as staticDRC gains derived from the selected dynamic range compression curvebased on broadband, wideband, and/or overall loudness levels over abroadband (or wideband) range or a plurality of frequency bands. As aresult, changes to the timbre effects that might be caused by applyingdifferent DRC gains in different frequency bands in other approaches canbe prevented.

In some embodiments, DRC gains in individual frequency bands arecontrolled with a broadband (or wideband) DRC gain determined based on abroadband (or wideband) loudness level. The DRC gains in the individualfrequency bands may operate around the broadband (or wideband) DRC gainlooked up in dynamic range compression curve based on the broadband (orwideband) loudness level, so that the DRC gains in the individualfrequency bands as time-averaged over a certain time interval (e.g.,longer than 5.3 milliseconds, 20 milliseconds, 50 milliseconds, 80milliseconds, 100 milliseconds, etc.) are the same as the broadband (orwideband) level as indicated in the dynamic range compression curve. Insome embodiments, loudness level fluctuations over short time intervalsrelative to the certain time interval deviating from the time-averagedDRC gains are permissible among channels and/or frequency bands. Theapproach ensures the application of correct multi-channel and/ormultiband time-averaged DRC gains as indicated in the dynamic rangecompression curve and prevents the DRC gains in the short time intervalsfrom deviating too much from such time-averaged DRC gains as indicatedin the dynamic range compression curve.

9. Volume Adjustment in Loudness Domain

Applying linear processing for volume adjustment to audio excitationsignals under other approaches that do not implement techniques asdescribed herein may cause low audible signal levels to become inaudible(e.g., falling below the frequency dependent hearing threshold of thehuman auditory system, etc.).

Under techniques as described herein, volume adjustments of audiocontent may be made or implemented in the loudness domain (e.g., with aSone representation, etc.), rather than the physical domain (e.g., witha dB_(SPL) representation, etc.). In some embodiments, loudness levelsin all bands are scaled with the same factor in the loudness domain forthe purpose of maintaining perceptual qualities and/or integrity ofloudness level relationships among all the bands at all volume levels.The volume adjustments based on setting and adjusting gains in theloudness domain as described herein may be converted back to, andimplemented through, non-linear processing in the physical domain (or inthe digital domain representing the physical domain) that appliesdifferent scaling factors to audio excitation signals in differentfrequency bands. The non-linear processing in the physical domain,converted from the volume adjustments in the loudness domain undertechniques as described herein, attenuates or enhances loudness levelsof audio content with DRC gains that prevent most or all of low audiblelevels in the audio content from becoming inaudible. In someembodiments, loudness level differences between loud and soft soundswithin a program are reduced—but not perceptually obliterated—with theseDRC gains to maintain the low audible signal levels above the hearingthreshold of the human auditory system. In some embodiments, at lowvolume levels, frequencies or frequency bands with excitation signallevels close to the threshold of hearing are less attenuated thus areperceptually audible, in order to maintain a similarity of spectralperception and perceived timbre, etc., across a large range of volumelevels.

Techniques as described herein may implement conversions (e.g., back andforth, etc.) between signal levels, gains, etc. in the physical domain(or in the digital domain representing the physical domain) and loudnesslevels, gains, etc., in the loudness domain. These conversions may bebased on forward and inverse versions of one or more non-linearfunctions (e.g., mappings, curves, piece-wise linear segments, look-uptables, etc.) constructed based on a model of the human auditory system.

10. Downmix Loudness Adjustment

In some embodiments, the audio content (152) is encoded in the encodedaudio signal (102) for a reference speaker configuration (e.g. asurround sound configuration, a 5.1 speaker configuration, etc.) thatcomprises a plurality of audio channels or speakers.

A recipient decoder that operates with a specific speaker configurationwith a smaller number of audio channels or speakers (e.g., a two-channelheadset configuration, etc.) is expected to downmix (e.g., with one ormore downmix equations, etc.) the audio content (152) as received fromthe encoded audio signal (102) from the plurality of audio channels inthe reference speaker configuration to the smaller number of audiochannels in the decoder's specific speaker configuration, perform gainadjustments to the downmix audio content, produce a downmix output soundoutput, etc.

Loudness levels as measured in the reference speaker configuration forindividual portions of the audio content (152) may be different fromloudness levels as measured in the specific speaker configuration suchas a two-channel configuration, etc., for the same individual portionsof the audio content (152). For example, if a portion of the audiocontent (152) before downmixing has a particular channel-dependent sounddistribution that concentrates in left front and right front channels ofthe reference speaker configuration, the loudness level of the sameportion of the audio content (152) after downmixing to the two-channelconfiguration may be higher or louder than the loudness level of thesame portion of the audio content (152) in the reference speakerconfiguration before downmixing. On the other hand, if a portion of theaudio content (152) before downmixing has a particular channel-dependentsound distribution that concentrates in other channels other than theleft front and right front channels of the reference speakerconfiguration, the loudness level of the same portion of the audiocontent (152) after downmixing to the two-channel configuration may belower or quieter than the loudness level of the same portion of theaudio content (152) in the reference speaker configuration beforedownmixing.

In some embodiments, an audio encoder (e.g., 150, etc.) as describedherein is configured to provide downmix related metadata (e.g.,comprising one or more downmix loudness parameters, etc.) to downstreamaudio decoders. The downmix related metadata from the audio encoder(150) can be used by downstream audio decoders to efficiently andconsistently perform (e.g., in real time, in near real time, etc.)downmix related gain adjustment operations, to allow the downstreamaudio decoders to produce relatively accurate actual target loudnesslevels in downmix sound outputs, to prevent inconsistencies in measuredloudness levels between the reference speaker configuration and thedecoders' specific speaker configurations, etc.

In some embodiments, the audio encoder (150) determines one or moredownmix parameters based at least in part on the audio content (152)encoded for a reference speaker configuration and a specific speakerconfiguration (e.g., a two-channel configuration, etc.) that isdifferent from the reference speaker configuration. In some embodiments,the downmix loudness parameters comprise one or more different sets ofdownmix loudness parameters for different types of downmixingoperations. The downmix loudness parameters may comprise a single set ofdownmix loudness parameters to be used by downstream audio decoders toperform a specific type of downmixing such as LtRt downmixing, LoRodownmixing, etc. The downmix loudness parameters may comprise two ormore sets of downmix loudness parameters to be used by downstream audiodecoders to perform any of two or more specific types of downmixing suchas LtRt downmixing, LoRo downmixing, etc. The downmix loudness datagenerated by the audio encoder (150) can carry one or more specificflags to indicate the presence of one or more sets of downmixingloudness parameters for one or more different types of downmixingoperations. The downmix loudness data may also include a preference flagto indicate which type of downmixing operations is preferred for theaudio content to be downmixed. The downmix loudness parameters may bedelivered to a downstream decoder as a part of metadata delivered in anencoded audio signal (102) that include the audio content (152) encodedfor the reference speaker configuration.

Examples of downmix loudness parameters as described herein may include,but are not limited to only: any of one or more downmix loudnessmetadata indicators, one or more downmix loudness data fields, etc. Inan example embodiment, the downmix loudness parameters may comprise anindicator (e.g., a one-bit data field denoted as “dmixloudoffste”, etc.)to indicate whether downmix loudness offset data exists, a data field(e.g., a 5-bit data field denoted as “5-bit dmixloudoffst”, etc.) toindicate a downmix loudness offset, etc. In some embodiments, one ormore instances of these indicators and data fields may be generated bythe audio encoder (150) for one or more different types of downmixingoperations.

In some embodiments, the “dmixloudoffste” field may be set to one (1),only when the encoded audio signal (102) carries audio data (e.g., audiosamples, etc.) for more than two channels; the “dmixloudoffst” field maybe carried if the “dmixloudoffste” field is set to one (1). In anexample in which the encoded audio signal (102) is an AC-3 or E-AC-3bitstream, etc., the “dmixloudoffste” field may be set to one (1), whenan audio coding mode (e.g., “acmod”, etc.) for the AC-3 or E-AC-3bitstream is set to a value greater than 2; such a value of the audiocoding mode indicates that the reference speaker configuration is amulti-channel speaker configuration comprising more than two audiochannels or speakers, and is neither a center-speaker-only configuration(e.g., with a value of 1 for “acmod”, etc.) nor aleft-front-and-right-front-only speaker configuration (e.g., with avalue of 2 for “acmod”, etc.).

The “dmixloudoffst” field may be used to indicate a difference betweenthe expected loudness of a downmix sound output from an (e.g., assumed,expected, etc.) audio decoder (e.g., an AC-3 decoder, an E-AC-3 decoder,etc.), and the measured loudness of such a downmix sound output, withsome or all of gain adjustments due to dialogue normalization, dynamicrange compression, fixed attenuation to protect against downmixoverload, etc., having been applied prior to performing measurementsthat yield the measured loudness. In some embodiments, the measuredloudness comprises one or more different sets of downmix loudnessmeasurements for one or more different types of the downmix sound outputwith one or more different sets of gain adjustments. In someembodiments, the audio encoder (150) generates one or more downmixesbased on one or more types (e.g., LtRt downmixing operation, LoRodownmixing operation, etc.) of downmixing operations. For example, theaudio encoder (150) can apply one or more different sets of downmixcoefficients/equations (e.g., LtRt downmix coefficients/equations, LoRodownmix coefficients/equations, etc.) to the audio content encoded forthe (e.g., multichannel, etc.) reference speaker configuration togenerate the one or more downmixes. In some embodiments, the audioencoder (150) may apply one or more different sets of gain adjustmentsto one or more of the downmixes to generate one or more different typesof the downmix sound output for loudness measurements. Examples of setsof gain adjustments include but are not limited only to: any of a set ofgain adjustments with null gain, a set of gain adjustments includinggain adjustments relating to dynamic range compression, a set of gainadjustments including gain adjustments relating to dialog normalization,a set of gain adjustments excluding gain adjustments relating to dynamicrange compression, a set of gain adjustments excluding gain adjustmentsrelating to dialog normalization, a set of gain adjustments includinggain adjustments relating to both dynamic range compression and dialognormalization, etc. After the one or more different types of the downmixsound output for loudness measurements are generated based on one ormore different combinations of the one or more different sets of gainadjustments and the one or more of the downmixes, the audio encoder(150) can generate the measured loudness by making one or more differentsets of downmix loudness measurements in any, some, or all of the one ormore different types of the downmix sound output. The measured loudnessmay be made by the audio encoder (150) in any of a variety of loudnessmeasurement standards (e.g., LKFS, LUFS, etc.), methods, tools, etc. Forthe purpose of illustration only, the measured loudness may berepresented by a LKFS value.

In some embodiments, the audio encoder (150) assumes that an audiodecoder (e.g., 100, etc.) described herein that is to decode the encodedaudio signal (102) with a dialog loudness level (e.g., “dialnorm”, etc.)is expected to apply a certain amount of attenuation (e.g., a differencebetween a reference loudness level and “dialnorm”, etc.) during decodingto align/adjust an output dialog loudness level of the downmixed soundoutput to the reference loudness level. For example, if the dialogloudness level “dialnorm” (e.g., as determined from the audio content(152) encoded for the reference speaker configuration such as a 5.1speaker configuration, etc.) has a value of −24 dB_(FS), and if thereference loudness level for the decoder's specific speakerconfiguration (e.g., a two-channel configuration to which the audiocontent (152) is to be downmixed, etc.) is −31 LKFS, then the audiodecoder (100) is expected to apply an attenuation of 7 dB toalign/adjust the output dialog loudness level to the reference loudnesslevel. In some embodiments, the reference loudness level (e.g., −31LKFS, etc.) for the decoder's specific speaker configuration representsthe expected loudness level (e.g., of a 2-channel downmix sound output,etc.).

In some embodiments, the “dmixloudoffst” field may be used by the audioencoder (150) to indicate any loudness deviation between (1) theexpected loudness level of the 2-channel downmix sound output and (2)the measured loudness level of the 2-channel downmix sound output, asmeasured after some or all of gain adjustments due to dialoguenormalization, dynamic range compression, fixed attenuation to protectagainst downmix overload, etc., have been applied. The “dmixloudoffst”field may comprise one or more instances for one or more different typesof downmixes after applying one or more different sets of gainadjustment, etc. The loudness deviation as indicated by the“dmixloudoffst” field may, but is not limited to only, include loudnesslevel differences caused by downmixing audio content from a referencespeaker configuration to a specific speaker configuration such as thetwo-channel configuration, etc. The loudness deviation corresponds to(e.g., represents the opposite to, etc.) a loudness offset that shouldbe applied by a decoder with a specific speaker configuration to whichthe audio content (152) is to be downmixed, in order to produce thereference loudness level in the downmix sound output.

In an example implementation, the “dmixloudoffst” field (e.g., aninstance thereof, etc.) may be set to a value in a (e.g., integer, etc.)value range from 0 to 30, corresponding to a range of loudness offsetsfrom −7.5 LKFS to +7.5 LKFS, in 0.5 LKFS steps. Additionally,optionally, or alternatively, a value of 31 for the “dmixloudoffst”field may be designated as a reserved value, and if present, may beinterpreted as a downmix loudness offset of 0 LKFS.

In some embodiments, a positive LKFS value (e.g., a value of 16, 17, . .. , 30 for the “dmixloudoffst” field) of the “dmixloudoffst” fieldindicates that the measured loudness level of the downmix sound outputis louder than the expected loudness level of the downmix sound outputby the magnitude of the indicated LKFS value. A negative LKFS value(e.g., a value of 0, 1, . . . , 15 for the “dmixloudoffst” field) of the“dmixloudoffst” field indicates that the measured loudness level of thedownmix sound output is quieter or no louder than the expected downmixloudness by the magnitude of the indicated LKFS value.

Some or all of the downmix loudness parameters may be (e.g.,additionally, optionally, alternatively, etc.) used by an audio decoder(e.g., 100, etc.) with a speaker configuration such as the specificspeaker configuration, etc., to control one or more audio processingoperations, algorithms, etc., that operate on the audio content (152) inthe encoded audio signal (102) in order to compensate for loudness leveldifferences—of individual portions of the audio content (152) in theencoded audio signal (102)—caused by downmixing the audio content (152)from the reference speaker configuration to the specific speakerconfiguration.

In some embodiments, an audio decoder (e.g., 100, etc.) described hereinis configured to decode (e.g., multi-channel, etc.) audio content fromthe encoded audio signal (102), extracts a dialog loudness level (e.g.,“dialnorm”, etc.) from loudness metadata delivered with the audiocontent, etc. The audio decoder (100) may be operating with a specificspeaker configuration (e.g., a two-channel configuration, etc.) that hasfewer audio channels than the reference speaker configuration to whichthe audio content corresponds.

In some embodiments, the audio decoder (100) uses one or more downmixequations to downmix the multi-channel audio content, as received froman encoded audio signal (102) for the reference speaker configurationfor which the multi-channel audio content was coded to the specificspeaker configuration at the audio decoder, performs one or more audioprocessing operations, algorithms, etc., on the audio content asdownmixed to generate a downmix sound output, etc. The audio decoder(100) may be capable of performing one or more different types ofdownmixing operations. The audio decoder (100) can be configured todetermine and perform a specific type (e.g., LtRt downmixing, LoRodownmixing, etc.) of downmixing operations based on one or more factors.These factors may include, but are not limited only to: one or more ofuser input that specifies a preference for a specific user-selected typeof downmixing operation, user input that specifies a preference for asystem-selected type of downmixing operations, capabilities of thespecific speaker configuration and/or the audio decoder (100),availability of downmix loudness metadata for the specific type ofdownmixing operation, any encoder-generated preference flag for a typeof downmixing operation, etc. In some embodiments, the audio decoder(100) may implement one or more precedence rules, may solicit furtheruser input, etc., to determine a specific type of downmixing operationif these factors conflict among themselves.

The one or more audio processing operations, algorithms, etc., include,but are not limited only to: a loudness attenuation operation thatapplies an amount of attenuation (e.g., a difference between a referenceloudness level and “dialnorm”, etc.) to align/adjust an output dialogloudness level of the downmixed sound output to the reference loudnesslevel based at least in part on the dialog loudness level (e.g.,“dialnorm”, etc.) and the reference loudness level (e.g., −31 LKFS,etc.). In some embodiments, the audio decoder (100) further performssome or all of gain adjustments due to dialogue normalization, dynamicrange compression, fixed attenuation to protect against downmixoverload, etc. In some embodiments, these gain adjustments maycorrespond to—e.g., may be the same or substantially the same as—thoseperformed by the audio encoder (150) in determining the measuredloudness level as previously described. One or more of these gainadjustments may be specific to the type (e.g., LtRt downmixing, LoRodownmixing, etc.) of downmixing operation performed by the audio decoder(100).

Additionally, optionally, or alternatively, in some embodiments, theaudio decoder (100) is configured to extract downmix loudness metadata(e.g., the “dmixloudoffste” field, the “dmixloudoffst” field, etc.), asa part of metadata delivered with the audio content, from the encodedaudio signal (102). In some embodiments, downmix loudness parameters inthe extracted downmix loudness metadata comprise one or more differentsets of downmix loudness parameters for different types of downmixingoperations as indicated to be present by one or more flags carried inthe downmix loudness metadata. In response to determining that the oneor more sets of downmix loudness parameters are present, the audiodecoder (100) can determine/select a set of downmix loudness parameters,among the one or more different sets of downmix loudness parameters,that corresponds to the specific type (e.g., LtRt downmixing, LoRodownmixing, etc.) of downmixing operation performed by the audio decoder(100). The audio decoder (100) determine (e.g., based on whether the“dmixloudoffste” field has a value of 1 or 0, etc.) whether there existsdownmix loudness offset data in the specific set of downmix loudnessparameters. In response to determining (e.g., based on that the“dmixloudoffste” field has a value of 1, etc.) that there exists downmixloudness offset data in the specific set of downmix loudness parameters,the audio decoder (100) performs a loudness adjustment operation basedon downmix loudness offsets in the downmix loudness metadata (e.g., the“dmixloudoffst” field in the same set of downmix loudness parameters,etc.) extracted with the audio content from the encoded audio signal(102). The downmix loudness metadata may comprise the “dmixloudoffst”field having one or more instances for one or more different types ofdownmixes after applying one or more different sets of gain adjustments,etc. Based on the actual downmixing operation and the actual set (e.g.,no gain adjustments, gain adjustments excluding those relating to DRC,gain adjustments including those relating to DRC, gain adjustmentsexcluding those relating to dialog normalization, gain adjustmentsincluding those relating to dialog normalization, gain adjustmentsincluding those relating to both dialog normalization and DRC, etc.) ofgain adjustments performed by the audio decoder (100), the audio decoder(100) can determine/select a specific instance of the one or moreinstances of the “dmixloudoffst” field in the downmix loudness metadata.

In response to determining that the “dmixloudoffst” field indicates apositive LKFS value (e.g., a value of 16, 17, . . . , 30 for the“dmixloudoffst” field), which means that the loudness level (as measuredby an upstream audio encoder such as 150, etc.) of the downmix soundoutput after applying some or all of gain adjustments due to dialoguenormalization, dynamic range compression, fixed attenuation to protectagainst downmix overload, etc., is louder than the expected loudnesslevel of the downmix sound output by the magnitude of the indicated LKFSvalue, the audio decoder (100) performs a further gain adjustment with anegative gain value having the magnitude of the indicated LKFS value,which lowers or adjust the loudness level of the downmix sound output tothe expected loudness (e.g., the reference loudness level, etc.).

In response to determining that the “dmixloudoffst” field indicates anegative LKFS value (e.g., a value of 1, 2, . . . , 15 for the“dmixloudoffst” field), which means that the (as measured by an upstreamaudio encoder such as 150, etc.) loudness level of the downmix soundoutput after applying some or all of gain adjustments due to dialoguenormalization, dynamic range compression, fixed attenuation to protectagainst downmix overload, etc., is quieter or no louder than theexpected loudness level of the downmix sound output by the magnitude ofthe indicated LKFS value, the audio decoder (100) performs a furthergain adjustment with a negative gain value having the magnitude of theindicated LKFS value, which increases or adjust the loudness level ofthe downmix sound output to the expected loudness (e.g., the referenceloudness level, etc.).

A negative LKFS value (e.g., a value of 0, 1, . . . , 15 for the“dmixloudoffst” field) of the “dmixloudoffst” field indicates that themeasured loudness level of the downmix sound output is quieter or nolouder than the expected downmix loudness by the magnitude of theindicated LKFS value. In some embodiments, if a negative LKFS value isindicated/signaled in the encoded audio signal (102) to a recipientdecoder, the recipient decoder (e.g., 150, etc.) can take actions toensure that any positive gain applied to the 2-channel downmix soundoutput to compensate for the negative LKFS value does not introduceclipping of loudness levels in the 2-channel downmix sound output.

The further gain adjustment based on the loudness offset indicated inthe downmix loudness metadata may, but is not limited only to, bespecific to the type of downmixing operation performed by the audiodecoder (100).

11. Additional Operations Related to Gains

Under techniques as described herein, other processing such as dynamicequalization, noise compensation, etc., can also be performed in theloudness (e.g., perceptual) domain, rather than in the physical domain(or a digital domain representing the physical domain).

In some embodiments, gains from some or all of a variety of processingsuch as DRC, equalization noise compensation, clip prevention, gainsmoothing, etc., may be combined in the same gains in the loudnessdomain and/or may be applied in parallel. In some other embodiments,gains from some or all of a variety of processing such as DRC,equalization noise compensation, clip prevention, gain smoothing, etc.,may be in separate gains in the loudness domain and/or may be applied inseries at least in part. In some other embodiments, gains from some orall of a variety of processing such as DRC, equalization noisecompensation, clip prevention, gain smoothing, etc., may be applied inorder.

12. Specific and Broadband (or Wideband) Loudness Levels

One or more audio processing elements, units, components, etc., such astransmission filters, auditory filterbank, synthesis filterbank,short-time-Fourier transform, etc., may be used by an encoder or decoderto perform audio processing operations as described herein.

In some embodiments, one or more transmission filters that model theouter and middle ear filtering of the human auditory system may be usedto filter an incoming audio signal (e.g., an encoded audio signal 102,audio content from a content provider, etc.). In some embodiments, anauditory filterbank may be used to model the frequency selectivity andfrequency spread of the human auditory system. Excitation signal levelsfrom some or all of these filters may be determined/calculated andsmoothed with frequency dependent time constants that are shortertowards higher frequencies to model the integration of energy in thehuman auditory system. Subsequently, a non-linear function (e.g.,relation, curve, etc.) between excitation signals and specific loudnesslevels may be used to obtain a profile of frequency-dependent specificloudness levels. A broadband (or wideband) loudness level can beobtained by integrating the specific loudness over frequency bands.

A straightforward (e.g., with equal weight to all frequency bands, etc.)summation/integration of specific loudness levels may work well forbroadband signals. However, such an approach may underestimate (e.g.,perceptual, etc.) loudness levels for narrowband signals. In someembodiments, specific loudness levels in different frequencies or indifferent frequency bands are given different weights.

In some embodiments, the auditory filterbanks and/or the transmissionfilters as mentioned above may be replaced by one or more Short-TimeFourier Transforms (STFT). Responses of the transmission filter andauditory filterbank may be applied in a Fast Fourier Transform (FFT)domain. In some embodiments, one or more inverse transmission filtersare used, for example, when one or more (e.g., forward, etc.)transmission filters are used in or before the conversion from thephysical domain (or in the digital domain representing the physicaldomain) to the loudness domain. In some embodiments, inversetransmission filters are not used, for example, when the STFT is used inplace of auditory filterbanks and/or transmission filters. In someembodiments, auditory filterbank are omitted; instead, one or morequadrature mirror filters (QMF) are used. In these embodiments, thespreading effect of the basilar membrane in the model of the humanauditory system may be omitted without significantly affecting theperformance of the audio processing operations as described herein.

Under techniques as described herein, different numbers of frequencybands (e.g., 20 frequency bands, 40 perceptual bands, etc.) may be usedin various embodiments. Additionally, optionally or alternatively,different bandwidth widths may also be used in various embodiments.

13. Individual Gains for Individual Subsets of Channels

In some embodiments, when a specific speaker configuration is amulti-channel configuration, an overall loudness levels may be obtainedby first summing excitation signals of all channels before theconversion from the physical domain (or in the digital domainrepresenting the physical domain) to the loudness domain. However,applying the same gains to all channels in the specific speakerconfiguration may not preserve spatial balance among the differentchannels (e.g., in terms of relative loudness levels between differentchannels, etc.) of the specific speaker configuration.

In some embodiments, to preserve the spatial balance such that relativeperceptual loudness levels among different channels may be optimally orcorrectly maintained, respective loudness levels and corresponding gainsobtained based on the respective loudness levels may be determined orcalculated per channel. In some embodiments, the corresponding gainsobtained based on the respective loudness levels do not equal the sameoverall gain; for example, each of some or all of the correspondinggains may equals to the overall gain plus a (e.g., channel-specific)small correction.

In some embodiments, to preserve the spatial balance, respectiveloudness levels and corresponding gains obtained based on the respectiveloudness levels may be determined or calculated per subset of channels.In some embodiments, the corresponding gains obtained based on therespective loudness levels do not equal the same overall gain; forexample, each of some or all of the corresponding gains may equals tothe overall gain plus a (e.g., channel-specific) small correction. Insome embodiments, a subset of channels may comprise two or more channels(e.g., a subset of channels comprising left front, right front, and lowfrequency effect (LFE); a subset of channels comprising left surroundand right surround; etc.) forming a proper subset of all channels in thespecific speaker configuration. Audio content for the subset of channelsmay constitute a submix of an overall mix carried in the encoded audiosignal (102). The channels within a submix can be applied with the samegains.

In some embodiments, to produce actual loudness (e.g., actuallyperceived, etc.) from a specific speaker configuration, one or morecalibration parameters may be used to relate signal levels in a digitaldomain to the corresponding physical (e.g., spatial pressure in terms ofdB_(SPL), etc.) levels in a physical domain represented by the digitaldomain. The one or more calibration parameters may be given values thatare specific to physical sound equipment in the specific speakerconfiguration.

14. Auditory Scene Analysis

In some embodiments, an encoder as described herein may implementcomputer-based auditory scene analysis (ASA) to detect auditory eventboundaries in audio content (e.g., encoded into the encoded audio signal102, etc.), generate one or more ASA parameters format the one or moreASA parameters as a part of an encoded audio signal (e.g., 102, etc.) tobe delivered to downstream devices (e.g., decoder 100, etc.). The ASAparameters may include but are not limited only to: any of parametersindicating locations of the auditory event boundaries, values of anauditory event certainty measure (as will be further explained below),etc.

In some embodiments, a (e.g., time-wise) location of an auditory eventboundary may be indicated in metadata encoded within the encoded audiosignal (102). Additionally, optionally, or alternatively, a (e.g.,time-wise) location of an auditory event boundary may be indicated(e.g., with a flag, a data field, etc.) in an audio data block and/orframe at which the location of the auditory event boundary is detected.

As used herein, an auditory event boundary refers to a point at which apreceding auditory event ends and/or a succeeding auditory event begins.Each auditory event occurs between two consecutive auditory eventboundaries.

In some embodiments, the encoder (150) is configured to detect auditoryevent boundaries by differences in specific loudness spectra between two(e.g., time-wise, etc.) consecutive audio data frames. Each of thespecific loudness spectra may comprise a spectrum of unsmoothed loudnesscomputed from a corresponding audio data frame of the consecutive audiodata frames.

In some embodiments, a specific loudness spectrum N[b, t] may benormalized to obtain a normalized specific loudness spectrum N_(NORM)[b, t] as shown in the following expression:

$\begin{matrix}{{N_{NORM}\left\lbrack {b,t} \right\rbrack} = \frac{N\left\lbrack {b,t} \right\rbrack}{\max\limits_{b}\left\{ {N\left\lbrack {b,t} \right\rbrack} \right\}}} & (1)\end{matrix}$where b indicates a band, t indicates a time or an audio data frameindex, and

$\max\limits_{b}\left\{ {N\left\lbrack {b,t} \right\rbrack} \right\}$is the maximum specific loudness level across all frequency bands.

Normalized specific loudness spectra may be subtracted from each otherand used to derive summed absolute differences, D[t], as shown in thefollowing expression:D[t]=Σ_(b) |N _(NORM)[b,t]−N _(NORM)[b,t−1]|  (2)

The summed absolute differences are mapped to an auditory eventcertainty measure A[t] with a value range of 0 to 1 as follows:

$\begin{matrix}{{A\lbrack t\rbrack} = \left\{ \begin{matrix}0 & {{D\lbrack t\rbrack} \leq D_{m\; i\; n}} \\\frac{{D\lbrack t\rbrack} - D_{m\; i\; n}}{D_{{ma}\; x} - D_{m\; i\; n}} & {D_{m\; i\; n} < {D\lbrack t\rbrack} < D_{{ma}\; x}} \\1 & {{D\lbrack t\rbrack} \geq D_{{ma}\; x}}\end{matrix} \right.} & (3)\end{matrix}$where D_(min) and D_(max) are minimum and maximum thresholds (e.g., userconfigurable, system configurable, set in relation to past valuedistribution of D[t] in the audio content, etc.).

In some embodiments, the encoder (150) is configured to detect anauditory event boundary (e.g., a specific t, etc.) when D[t] (e.g., atthe specific t, etc.) rises above D_(min).

In some embodiments, a decoder (e.g., 100, etc.) as described hereinextracts the ASA parameters from an encoded audio signal (e.g., 102,etc.) and use the ASA parameters to prevent unintentional boosting ofsoft sounds and/or unintentional cutting of loud sounds that causeperceptual distortions of auditory events.

The decoder (100) may be configured to reduce or prevent unintentionaldistortions of auditory events by ensuring that within an auditory eventthe gain is more nearly constant and by confining much of the gainchange to the neighborhood of an auditory event boundary. For example,the decoder (100) may be configured to use a relatively small timeconstant (e.g., comparable with or shorter than a minimum duration ofauditory events, etc.) in response to a gain change in an attack (e.g.,loudness level increasing, etc.) at an auditory event boundary.Accordingly, the gain change in the attack can be implemented by thedecoder (100) relatively rapidly. On the other hand, the decoder (100)may be configured to use a relatively long time constant relative to aduration of an auditory event in response to a gain change in a release(e.g., loudness level decreasing, etc.) in an auditory event.Accordingly, the gain change in the release can be implemented by thedecoder (100) relatively slowly so that sounds that ought to appearconstant or to decay gradually may not be audibly or perceptuallydisturbed. The quick response in an attack at an auditory event boundaryand the slow response in a release in an auditory event allow a fastperception of an arrival of the auditory event and preserve perceptualqualities and/or integrity during the auditory event—which comprisesloud and soft sounds linked by specific loudness level relationshipsand/or specific time relationships—such as a piano chord, etc.

In some embodiments, auditory events and auditory event boundariesindicated by the ASA parameters are used by the decoder (100) to controlgain changes in one, two, some or all of the channels in a specificspeaker configuration at the decoder (100).

15. Loudness Level Transitions

Loudness level transitions may occur, for example, between two programs,between a program and a loud commercial, etc. In some embodiments,decoder (100) is configured to maintain a histogram of instantaneousloudness levels based on past audio content (e.g., received from theencoded audio signal 102, for the past 4 seconds, etc.). Over a timeinterval from before a loudness level transition to after the loudnesslevel transition, two areas with heightened probabilities may berecorded in the histogram. One of the areas centers around a previousloudness level, whereas the other the areas centers around a newloudness level.

The decoder (100) may dynamically determine a smoothed loudness level asthe audio content is being processed, and determine a corresponding bin(e.g., a bin of instantaneous loudness levels that include the samevalue as the smoothed loudness level, etc.) of the histogram based onthe smoothed loudness level. The decoder (100) is further configured tocompare a probability at the corresponding bin with a threshold (e.g.,6%, 7%, 7.5%, etc.), where the total area (e.g. the sum of all bins) ofthe histogram curve represents a probability of 100%. The decoder can beconfigured to detect the occurrence of the loudness level transition bydetermining that the probability at the corresponding bin falls belowthe threshold. In response, the decoder (100) may be configured toselect a relatively small time constant to adapt relatively fast to thenew loudness level. Consequently, time durations of loud (or soft)onsets within loudness level transitions can be reduced.

In some embodiments, the decoder (100) uses a silence/noise gate toprevent low instantaneous loudness levels from entering into thehistogram and becoming a high probability bin in the histogram.Additionally, optionally or alternatively, the decoder (100) may beconfigured to use the ASA parameters to detect auditory events to beincluded in the histogram. In some embodiments, the decoder (100) maydetermine time-dependent values of a time-averaged auditory eventcertainty measure Ā[t] from the ASA parameters. In some embodiments, thedecoder (100) may determine time-dependent values of an (e.g.,instantaneous, etc.) auditory event certainty measure A[t] from the ASAparameters and compute values of a time-averaged auditory eventcertainty measure Ā[t] based on the time-dependent values of an (e.g.,instantaneous, etc.) auditory event certainty measure A[t] from the ASAparameters, etc. The decoder (100) may be configured to exclude loudnesslevels from entering the histogram if the time-averaged auditory eventcertainty measure Ā[t] contemporaneous with the loudness levels fallbelow a histogram inclusion threshold value (e.g., 0.1, 0.12, etc.).

In some embodiments, for (e.g., instantaneous, etc.) loudness levels(e.g., corresponding Ā[t] values are above the histogram inclusionthreshold value, etc.) permitted to be included in the histogram, theloudness levels are assigned weights that are the same as, proportionalto, etc., time dependent values of the time-averaged auditory eventcertainty measure Ā[t] contemporaneous with the loudness levels. As aresult, loudness levels near an auditory event boundary have moreinfluence on the histogram (e.g., A[t] has relatively large values,etc.) than other loudness levels that are not near an auditory eventboundary.

16. Reset

In some embodiments, an encoder as described herein (e.g., 150, etc.) isconfigured to detect reset events and include indications of the resetevents in an encoded audio signal (e.g., 102, etc.) generated by theencoder (150). In a first example, the encoder (150) detects a resetevent in response to determining that there occurs a continuous (e.g.,250 milliseconds, configurable by a system and/or a user, etc.) periodof relative silence. In a second example, the encoder (150) detects areset event in response to determining that there occurs a largeinstantaneous drop in excitation level across all frequency bands. In athird example the encoder is provided with input (e.g. metadata, userinput, system controlled, etc.) where transitions in content (e.g.program start/end, scene change, etc.) occur that require a reset.

In some embodiments, a decoder as described herein (e.g., 100, etc.)implements a reset mechanism that can be used to instantaneously speedup gain smoothing. The reset mechanism is useful and may be invoked whenswitching between channels or audiovisual inputs occurs.

In some embodiments, the decoder (100) can be configured to determinewhether a reset event occurs by determining whether there occurs acontinuous (e.g., 250 milliseconds, configurable by a system and/or auser, etc.) period of relative silence, whether there occurs a largeinstantaneous drop in excitation level across all frequency bands, etc.

In some embodiments, the decoder (100) can be configured to determinethat a reset event occurs in response to receiving an indication (e.g.,of the reset event, etc.) that was provided in an encoded audio signal(102) by an upstream encoder (e.g., 150, etc.).

The reset mechanism may be caused to issue a reset when the decoder(100) determining that a reset event occurs. In some embodiments, thereset mechanism is configured to use a slightly more aggressive cutbehavior of the DRC compression curve to prevent hard onsets (e.g., of aloud program/channel/audiovisual source, etc.). Additionally,optionally, or alternatively, the decoder (100) may be configured toimplement safeguards to recover gracefully when the decoder (100)detects that a reset is falsely triggered.

17. Encoder-Provided Gains

In some embodiments, the audio encoder can be configured to compute oneor more sets of gains (e.g., DRC gains, etc.) for individual portions(e.g., audio data blocks, audio data frames, etc.) of the audio contentto be encoded into the encoded audio signal. The sets of gains generatedby the audio encoder may comprise one or more of: a first set of gainscomprising a single broadband (or wideband) gain for all channels (e.g.,left front, right front, low frequency effect or LFE, center, leftsurround, right surround, etc.); a second set of gains comprisingindividual broadband (or wideband) gains for individual subsets ofchannels; a third set of gains comprising individual broadband (orwideband) gains for individual subsets of channels and for each of afirst number (e.g., two, etc.) of individual bands (e.g., two bands ineach channel, etc.); a fourth set of gains comprising individualbroadband (or wideband) gains for individual subsets of channels and foreach of a second number (e.g., four, etc.) of individual bands (e.g.,four bands in each channel, etc.); etc. A subset of channels asdescribed herein may be one of a subset comprising left front, rightfront and LFE channels, a subset comprising a center channel, a subsetcomprising left surround and right surround channels, etc.

In some embodiments, the audio encoder is configured to transmit one ormore portions (e.g., audio data blocks, audio data frames, etc.) of theaudio content and one or more sets of gains computed for the one or moreportions of the audio content in a time-synchronous manner. An audiodecoder that receives the one or more portions of the audio content canselect and apply a set of gains among the one or more sets of gains withlittle or no delay. In some embodiments, the audio encoder can implementsub-framing techniques under which the one or more sets of gains arecarried (e.g., with differential coding, etc.) in one or more sub-framesas illustrated in FIG. 4. In an example, the sub-frames may be encodedwithin the audio data blocks or audio data frames for which the gainsare computed. In another example, the sub-frames may be encoded withinaudio data blocks or audio data frames preceding the audio data blocksor audio data frames for which the gains are computed. In anothernon-limiting example, the sub-frames may be encoded within audio datablocks or audio data frames within a certain time from the audio datablocks or audio data frames for which the gains are computed. In someembodiments, Huffman and differential coding may be used to populateand/or compress the sub-frames that carry the sets of gains.

18. Example System and Process FlowS

FIG. 5 illustrates an example codec system in a non-limiting exampleembodiment. A content creator, which may be a processing unit in anaudio encoder such as 150, etc., is configured to provide audio content(“Audio”) to an encoder unit (“NGC Encoder”). The encoder unit formatsthe audio content into audio data blocks and/or frames and encodes theaudio data blocks and/or frames into an encoded audio signal. Thecontent creator is also configured to establish/generate one or moredialog loudness levels (“dialnorm”) of one or more programs,commercials, etc., in the audio content and one or more dynamic rangecompression curve identifiers (“Compression curve IDs”). The contentcreator may determine the dialog loudness levels from one or moredialogue audio tracks in the audio content. The dynamic rangecompression curve identifiers may be selected based at least in part onuser input, system configuration parameters, etc. The content creatormay be a person (e.g. artist, audio engineer, etc.) using tools togenerate the audio content and dialnorm.

Based on the dynamic range compression curve identifiers, the encoder(150) generates one or more DRC parameter sets including but not limitedto corresponding reference dialogue loudness levels (“Reference levels”)for a plurality of playback environments supported by the one or moredynamic range compression curves. These DRC parameter sets may beencoded in-band with the audio content, out-of-band with the audiocontent, etc., in metadata of the encoded audio signal. Operations suchas compression, formatting multiplexing (“MUX”), etc., may be performedas a part of generating the encoded audio signal that may be deliveredto an audio decoder such as 100, etc. An encoded audio signal may beencoded with a syntax that supports carriage of audio data elements, DRCparameter sets, reference loudness levels, dynamic range compressioncurves, functions, lookup tables, Huffman codes used in compression,sub-frames, etc. In some embodiments, the syntax allows an upstreamdevice (e.g., an encoder, a decoder, a transcoder, etc.) to transmitgains to a downstream device (e.g., a decoder, a transcoder, etc.). Insome embodiments, the syntax used to encode data into and/or decode thedata from an encoded audio signal is configured to support backwardcompatibility such that a device that relies on gains computed by anupstream device may optionally continue to do so.

In some embodiments, the encoder (150) computes one, two or more sets ofgains (e.g., DRC gains, gain smoothing, with appropriate referencedialogue loudness levels, etc.) for the audio content. The sets forgains may be provided with the one or more dynamic range compressioncurves in the metadata encoded with the audio content into the encodedaudio signal. A first set of gains may correspond to a broadband (orwideband) gain for all channels in a (e.g., default, etc.) speakerconfiguration or profile. A second set of gains may correspond to abroadband (or wideband) gain for each of the all channels in the speakerconfiguration or profile. A third set of gains may correspond to abroadband (or wideband) gain for each of two bands in each of the allchannels in the speaker configuration or profile. A fourth set of gainsmay correspond to a broadband (or wideband) gain for each of four bandsin each of the all channels in the speaker configuration or profile. Insome embodiments, the sets of gains computed for a speaker configurationmay be transmitted with a (e.g., parameterized, etc.) dynamic rangecompression curve for the speaker configuration in the metadata. In someembodiments, the sets of gains computed for a speaker configuration mayreplace a (e.g., parameterized, etc.) dynamic range compression curvefor the speaker configuration in the metadata. Additional speakerconfigurations or profiles may be supported under techniques asdescribed herein.

The decoder (100) is configured to extract the audio data blocks and/orframes and the metadata from the encoded audio signals, for example,through operations such as decompression, deformatting, demultiplexing(“DEMUX”), etc. The extracted audio data blocks and/or frames may bedecoded by a decoder unit (“NGC Decoder”) into audio data elements orsamples. The decoder (100) is further configured to determine a profilefor a specific playback environment at the decoder (100), in which theaudio content is to be rendered, and select a dynamic range compressioncurve from the metadata extracted from the encoded audio signal. Adigital audio processing unit (“DAP”) is configured to apply DRC andother operations on the audio data elements or samples for the purposeof generating audio signals that drive audio channels in the specificplayback environment. The decoder (100) can calculate and apply DRCgains based on loudness levels determined from audio data blocks orframes and the selected dynamic range compression curve. The decoder(100) can also adjust the output dialogue loudness level based on areference dialogue loudness level associated with the selected dynamicrange compression curve and the dialogue loudness levels in the metadataextracted from the encoded audio signal. The decoder (100) cansubsequently apply gain limiter that is specific to a playback scenarioas related to the audio content and the specific playback environment.Thus, the decoder (100) can render/play the audio content as tailored tothe playback scenario.

FIG. 6A through FIG. 6D illustrate example process flows. In someembodiments, one or more computing devices or units in a mediaprocessing system may perform this process flow.

FIG. 6A illustrates an example process flow that may be implemented byan audio decoder as described herein. In block 602 of FIG. 6A, a firstdevice (e.g., an audio decoder 100 of FIG. 1A, etc.) receives an audiosignal that comprises audio content and definition data for one or moredynamic range compression curves.

In block 604, the first device determines a specific playbackenvironment.

In block 606, the first device establishes a specific dynamic rangecompression curve for the specific playback environment based on thedefinition data for the one or more dynamic range compression curvesextracted from the audio signal.

In block 608, the first device performs one or more dynamic rangecontrol (DRC) operations on one or more portions of the audio contentextracted from the audio signal. The one or more DRC operations beingbased at least in part on one or more DRC gains obtained from thespecific dynamic range compression curve.

In an embodiment, the definition data for the one or more dynamic rangecompression curves comprises one or more of attack times, release times,or reference loudness levels related to at least one of the one or moredynamic range compression curves.

In an embodiment, the first device is further configured to perform:computing one or more loudness levels for the one or more portions ofthe audio content; determining the one or more DRC gains based on thespecific dynamic range compression curve and the one or more loudnesslevels for the one or more portions of the audio content; etc.

In an embodiment, at least one of the loudness levels computed for theone or more portions of the audio content is one or more of specificloudness levels relating to one or more frequency bands, broadbandloudness levels across a broadband range, wideband loudness levelsacross a wideband range, broadband loudness levels across a plurality offrequency bands, wideband loudness levels across a plurality offrequency bands, etc.

In an embodiment, at least one of the loudness levels computed for theone or more portions of the audio content is one or more ofinstantaneous loudness levels or loudness levels smoothed over one ormore time intervals.

In an embodiment, the one or more operations comprise one or moreoperations related to one or more of adjusting dialog loudness levels,gain smoothing, gain limiting, dynamic equalization, noise compensation,etc.

In an embodiment, the first device is further configured to perform:extracting one or more dialogue loudness levels from the encoded audiosignal; adjusting the one or more dialogue loudness levels to one ormore reference dialogue loudness levels; etc.

In an embodiment, the first device is further configured to perform:extracting one or more auditory scene analysis (ASA) parameters from theencoded audio signal; changing one or more time constants used insmoothing gains applied to the audio content, the gains relating to oneor more of the one or more DRC gains, gain smoothing, or gain limiting;etc.

In an embodiment, the first device is further configured to perform:determining that a reset event occurs in the one or more portions of theaudio content based on an indication of a reset event, the indication ofthe reset being extracted from the encoded audio signal; in response todetermining that the reset event occurs in the one or more portions ofthe audio content, taking one or more actions on one or more gainsmoothing operations being performed at a time of determining that thereset event occurs in the one or more portions of the audio content;etc.

In an embodiment, the first device is further configured to perform:maintaining a histogram of instantaneous loudness levels, the histogrambeing populated by instantaneous loudness levels computed from a timeinterval in the audio content; determining whether a specific loudnesslevel is above a threshold in a high probability area of the histogram,the specific loudness level being computed from a portion of the audiocontent; in response to determining that the specific loudness level isabove the threshold in the high probability area of the histogram,performing: determining that a loudness transition occurs, shortening atime constant used in gain smoothing to speed up the loudnesstransition, etc.; etc.

FIG. 6B illustrates an example process flow that may be implemented byan audio encoder as described herein. In block 652 of FIG. 6B, a seconddevice (e.g., an audio encoder 150 of FIG. 1B, etc.) receives audiocontent in a source audio format.

In block 654, the second device retrieves definition data for one ormore dynamic range compression curves.

In block 656, the second device generates an audio signal that comprisesthe audio content and the definition data for the one or more dynamicrange compression curves.

In an embodiment, the second device is further configured to perform:determining one or more identifiers for the one or more dynamic rangecompression curves; retrieving the definition data for the one or moredynamic range compression curves from a reference data repository basedon the one or more identifiers; etc.

In an embodiment, the second device is further configured to perform:computing one or more dialogue loudness levels for the one or moreportions of the audio content; encoding the one or more dialogueloudness levels with the one or more portions of the audio content intothe encoded audio signal; etc.

In an embodiment, the second device is further configured to perform:performing auditory event scene (ASA) on the one or more portions of theaudio content; generating one or more ASA parameters based on results ofthe ASA on the one or more portions of the audio content; encoding theone or more ASA parameters with the one or more portions of the audiocontent into the encoded audio signal; etc.

In an embodiment, the second device is further configured to perform:determining that one or more reset events occur in the one or moreportions of the audio content; encoding one or more indications of theone or more reset events with the one or more portions of the audiocontent into the encoded audio signal; etc.

In an embodiment, the second device is further configured to encode theone or more portions of the audio content into one or more of audio dataframes or audio data blocks.

In an embodiment, a first DRC gain of the one or more DRC gains appliesto each channel in a first proper subset in a set of all channels in aspecific speaker configuration that corresponds to the specific playbackenvironment, whereas a second different DRC gain of the one or more DRCgains applies to each channel in a second proper subset in the set ofall channels in the specific speaker configuration that corresponds tothe specific playback environment.

In an embodiment, a first DRC gain of the one or more DRC gains appliesto a first frequency band, whereas a second different DRC gain of theone or more DRC gains applies to a second different frequency band.

In an embodiment, the one or more portions of the audio content compriseone or more of audio data frames or audio data blocks. In an embodiment,the encoded audio signal is a part of an audiovisual signal.

In an embodiment, the one or more DRC gains are defined in a loudnessdomain.

FIG. 6C illustrates an example process flow that may be implemented byan audio encoder as described herein. In block 662 of FIG. 6C, a thirddevice (e.g., an audio encoder 150 of FIG. 1B, etc.) generates audiocontent coded for a reference speaker configuration.

In block 664, the second device downmixes the audio content coded forthe reference speaker configuration to downmix audio content coded for aspecific speaker configuration.

In block 666, the second device performs one or more gain adjustments onindividual portions of the downmix audio content coded for the specificspeaker configuration.

In block 668, the second device performs loudness measurements on theindividual portions of the downmix audio content.

In block 670, the second device generates an audio signal that comprisesthe audio content coded for the reference speaker configuration anddownmix loudness metadata created based at least in part on the loudnessmeasurements on the individual portions of the downmix audio content;

In an embodiment, the loudness measurements on the individual portionsof the downmix audio content are performed after the one or more gainadjustments are applied to the individual portions of the downmix audiocontent. In some embodiments, the loudness measurements are based on aLoudness-K-weighted-Full-Scale (LKFS) standard. In some otherembodiments, the loudness measurements are based on a loudness standardother than a Loudness-K-weighted-Full-Scale (LKFS) standard.

In an embodiment, the audio content coded for the reference speakerconfiguration is downmixed to the downmix audio content coded for thespecific speaker configuration based on one or more types of downmixingoperations; the loudness measurements on the individual portions of thedownmix audio content include loudness measurements on the individualportions of the downmix audio content relating to each of the one ormore types of downmixing operations.

In an embodiment, the third device is further configured to prevent thedownmixed audio content for the specific speaker configuration frombeing encoded in the audio signal.

FIG. 6D illustrates an example process flow that may be implemented byan audio decoder as described herein. In block 682 of FIG. 6D, a fourthdevice (e.g., an audio decoder 100 of FIG. 1A, etc.) operating with aspecific speaker configuration receives an audio signal that comprisesaudio content coded for a reference speaker configuration and downmixloudness metadata.

In block 684, the first device downmixes the audio content coded for thereference speaker configuration to downmix audio content coded for thespecific speaker configuration.

In block 686, the first device performs one or more gain adjustments onindividual portions of the downmix audio content coded for the specificspeaker configuration. The one or more gain adjustments are not based onthe downmix loudness metadata; and correspond to one or more gainadjustments performed by an upstream audio encoder, before generatingthe downmix loudness metadata by the upstream audio encoder.

In block 688 the first device performs one or more additional gainadjustments on the individual portions of the downmix audio contentcoded for the specific speaker configuration, the one or more additionalgain adjustment being based on the downmix loudness metadata.

In an embodiment, the first device is further configured to perform:determining a specific type of downmixing operation based on one or moreselection factors; applying the specific type of downmixing operation indownmixing the audio content coded for the reference speakerconfiguration to the downmix audio content coded for the specificspeaker configuration; determining, from one or more sets of downmixingloudness parameters in the downmix loudness metadata, a specific set ofdownmix loudness parameters to which the specific type of downmixingoperation correspond; and performing the one or more additional gainadjustments on the individual portions of the downmix audio contentcoded for the specific speaker configuration based at least in part onthe specific set of downmix loudness parameters.

In an embodiment, the one or more gain adjustments do not produce anexpected loudness in a downmix sound output for at least one individualportion of the one or more individual portions of the downmix audiocontent, wherein the one or more additional gain adjustments areperformed to produce an expected loudness in a downmix sound output forthe at least one individual portion of the one or more individualportions of the downmix audio content.

In an embodiment, the reference speaker configuration is a surroundspeaker configuration, and wherein the specific speaker configuration isa two-channel configuration.

In an embodiment, the audio content coded for the reference speakerconfiguration is downmixed to the downmix audio content coded for thespecific speaker configuration based on one or more downmix equations.

In an embodiment, the downmix loudness metadata comprises one or moresets of downmix loudness parameters, each set of the two or more sets ofdownmix loudness parameters corresponding to an individual type ofdownmixing operation among one or more types of downmix operations towhich the one or more sets of downmix loudness parameters correspond.

In an embodiment, the one or more types of downmixing operationscomprise at least one of LtRt dowmixing operation or LoRo downmixingoperation.

In an embodiment, the one or more gain adjustments comprise at least onegain adjustment relating to one or more of dialogue normalization,dynamic range compression, or fixed attenuation to protect againstdownmix overload.

In an embodiment, the one or more gain adjustments use different gainadjustment parameter values for at least two different portions of theindividual portions of the audio content.

In an embodiment, the downmix loudness metadata represents a part ofoverall audio metadata encoded in the audio signal. In an embodiment,the downmix loudness metadata comprises a data field to indicate adownmix loudness offset. In an embodiment, the encoded audio signal is apart of an audiovisual signal.

In an embodiment, an apparatus comprising a processor and configured toperform any one of the methods as described herein.

In an embodiment, a non-transitory computer readable storage medium,comprising software instructions, which when executed by one or moreprocessors cause performance of any one of the methods as describedherein. Note that, although separate embodiments are discussed herein,any combination of embodiments and/or partial embodiments discussedherein may be combined to form further embodiments.

19. Implementation Mechanisms—Hardware Overview

According to one embodiment, the techniques described herein areimplemented by one or more special-purpose computing devices. Thespecial-purpose computing devices may be hard-wired to perform thetechniques, or may include digital electronic devices such as one ormore application-specific integrated circuits (ASICs) or fieldprogrammable gate arrays (FPGAs) that are persistently programmed toperform the techniques, or may include one or more general purposehardware processors programmed to perform the techniques pursuant toprogram instructions in firmware, memory, other storage, or acombination. Such special-purpose computing devices may also combinecustom hard-wired logic, ASICs, or FPGAs with custom programming toaccomplish the techniques. The special-purpose computing devices may bedesktop computer systems, portable computer systems, handheld devices,networking devices or any other device that incorporates hard-wiredand/or program logic to implement the techniques.

For example, FIG. 7 is a block diagram that illustrates a computersystem 700 upon which an embodiment of the invention may be implemented.Computer system 700 includes a bus 702 or other communication mechanismfor communicating information, and a hardware processor 704 coupled withbus 702 for processing information. Hardware processor 704 may be, forexample, a general purpose microprocessor.

Computer system 700 also includes a main memory 706, such as a randomaccess memory (RAM) or other dynamic storage device, coupled to bus 702for storing information and instructions to be executed by processor704. Main memory 706 also may be used for storing temporary variables orother intermediate information during execution of instructions to beexecuted by processor 704. Such instructions, when stored innon-transitory storage media accessible to processor 704, rendercomputer system 700 into a special-purpose machine that isdevice-specific to perform the operations specified in the instructions.

Computer system 700 further includes a read only memory (ROM) 708 orother static storage device coupled to bus 702 for storing staticinformation and instructions for processor 704. A storage device 710,such as a magnetic disk or optical disk, is provided and coupled to bus702 for storing information and instructions.

Computer system 700 may be coupled via bus 702 to a display 712, such asa liquid crystal display (LCD), for displaying information to a computeruser. An input device 714, including alphanumeric and other keys, iscoupled to bus 702 for communicating information and command selectionsto processor 704. Another type of user input device is cursor control716, such as a mouse, a trackball, or cursor direction keys forcommunicating direction information and command selections to processor704 and for controlling cursor movement on display 712. This inputdevice typically has two degrees of freedom in two axes, a first axis(e.g., x) and a second axis (e.g., y), that allows the device to specifypositions in a plane.

Computer system 700 may implement the techniques described herein usingdevice-specific hard-wired logic, one or more ASICs or FPGAs, firmwareand/or program logic which in combination with the computer systemcauses or programs computer system 700 to be a special-purpose machine.According to one embodiment, the techniques herein are performed bycomputer system 700 in response to processor 704 executing one or moresequences of one or more instructions contained in main memory 706. Suchinstructions may be read into main memory 706 from another storagemedium, such as storage device 710. Execution of the sequences ofinstructions contained in main memory 706 causes processor 704 toperform the process steps described herein. In alternative embodiments,hard-wired circuitry may be used in place of or in combination withsoftware instructions.

The term “storage media” as used herein refers to any non-transitorymedia that store data and/or instructions that cause a machine tooperation in a specific fashion. Such storage media may comprisenon-volatile media and/or volatile media. Non-volatile media includes,for example, optical or magnetic disks, such as storage device 710.Volatile media includes dynamic memory, such as main memory 706. Commonforms of storage media include, for example, a floppy disk, a flexibledisk, hard disk, solid state drive, magnetic tape, or any other magneticdata storage medium, a CD-ROM, any other optical data storage medium,any physical medium with patterns of holes, a RAM, a PROM, and EPROM, aFLASH-EPROM, NVRAM, any other memory chip or cartridge.

Storage media is distinct from but may be used in conjunction withtransmission media. Transmission media participates in transferringinformation between storage media. For example, transmission mediaincludes coaxial cables, copper wire and fiber optics, including thewires that comprise bus 702. Transmission media can also take the formof acoustic or light waves, such as those generated during radio-waveand infra-red data communications.

Various forms of media may be involved in carrying one or more sequencesof one or more instructions to processor 704 for execution. For example,the instructions may initially be carried on a magnetic disk or solidstate drive of a remote computer. The remote computer can load theinstructions into its dynamic memory and send the instructions over atelephone line using a modem. A modem local to computer system 700 canreceive the data on the telephone line and use an infra-red transmitterto convert the data to an infra-red signal. An infra-red detector canreceive the data carried in the infra-red signal and appropriatecircuitry can place the data on bus 702. Bus 702 carries the data tomain memory 706, from which processor 704 retrieves and executes theinstructions. The instructions received by main memory 706 mayoptionally be stored on storage device 710 either before or afterexecution by processor 704.

Computer system 700 also includes a communication interface 718 coupledto bus 702. Communication interface 718 provides a two-way datacommunication coupling to a network link 720 that is connected to alocal network 722. For example, communication interface 718 may be anintegrated services digital network (ISDN) card, cable modem, satellitemodem, or a modem to provide a data communication connection to acorresponding type of telephone line. As another example, communicationinterface 718 may be a local area network (LAN) card to provide a datacommunication connection to a compatible LAN. Wireless links may also beimplemented. In any such implementation, communication interface 718sends and receives electrical, electromagnetic or optical signals thatcarry digital data streams representing various types of information.

Network link 720 typically provides data communication through one ormore networks to other data devices. For example, network link 720 mayprovide a connection through local network 722 to a host computer 724 orto data equipment operated by an Internet Service Provider (ISP) 726.ISP 726 in turn provides data communication services through the worldwide packet data communication network now commonly referred to as the“Internet” 728. Local network 722 and Internet 728 both use electrical,electromagnetic or optical signals that carry digital data streams. Thesignals through the various networks and the signals on network link 720and through communication interface 718, which carry the digital data toand from computer system 700, are example forms of transmission media.

Computer system 700 can send messages and receive data, includingprogram code, through the network(s), network link 720 and communicationinterface 718. In the Internet example, a server 730 might transmit arequested code for an application program through Internet 728, ISP 726,local network 722 and communication interface 718.

The received code may be executed by processor 704 as it is received,and/or stored in storage device 710, or other non-volatile storage forlater execution.

20. Equivalents, Extensions, Alternatives and Miscellaneous

In the foregoing specification, embodiments of the invention have beendescribed with reference to numerous specific details that may vary fromimplementation to implementation. Thus, the sole and exclusive indicatorof what is the invention, and is intended by the applicants to be theinvention, is the set of claims that issue from this application, in thespecific form in which such claims issue, including any subsequentcorrection. Any definitions expressly set forth herein for termscontained in such claims shall govern the meaning of such terms as usedin the claims. Hence, no limitation, element, feature, feature,advantage or attribute that is not expressly recited in a claim shouldlimit the scope of such claim in any way. The specification and drawingsare, accordingly, to be regarded in an illustrative rather than arestrictive sense.

What is claimed is:
 1. A method, comprising: receiving an encoded audiobitstream, the encoded audio bitstream including audio data and metadataaudio signal including one or more sets of downmix loudness parameters;determining if a first downmix loudness parameter indicates whetherdownmix loudness offset data exists, and if the downmix loudness offsetdata does exist, adjusting a difference between an expected loudness ofa downmix audio signal and a measured loudness of the downmix audiosignal based upon a second downmix loudness parameter; and wherein thesecond downmix loudness parameter indicates whether there is adifference between the expected loudness of a 2-channel downmix outputand an actual measured loudness.
 2. The method of claim 1 wherein, thefirst downmix loudness parameter is a one-bit data field denoted as“dmixloudoffste” and the second downmix loudness parameter is a five-bitfield denoted as “dmixloudoffst.”
 3. The method of claim 2, wherein thedmixloudoffst indicates whether there is a difference between theexpected loudness of the 2-channel downmix output from an AC-3 orEnhanced AC-3 decoder, and the actual measured loudness, with all gainadjustments due to dialogue normalization, dynamic range compressionand/or fixed attenuation to protect against downmix overload having beenapplied prior to measurement.
 4. The method of claim 1, wherein aprogram loudness data conveys data about program loudness, includingmeasured loudness values for the program, and indication as to whetherthe loudness of the program has been corrected prior to AC-3 or EnhancedAC-3 encoding.
 5. An audio processing apparatus comprising: a buffer forstoring at least a portion of an encoded audio bitstream, the encodedaudio bitstream including audio data and metadata including one or moresets of downmix loudness parameters; a demultiplexer for parsing theportion of the encoded audio bitstream; and an audio decoder fordecoding the audio data, wherein the metadata includes a first downmixloudness parameter indicating whether downmix loudness offset dataexists, and if the downmix loudness offset data does exist, a seconddownmix loudness parameter indicating whether there is a differencebetween an expected loudness of a 2-channel downmix of the audio dataand an actual measured loudness of the 2-channel downmix of the audiodata.
 6. The audio processing apparatus of claim 5, wherein, a firstdownmix loudness parameter is a one-bit data field denoted as“dmixloudoffste” and the second downmix loudness parameter is a five-bitfield denoted as “dmixloudoffst.”
 7. The audio processing apparatus ofclaim 5, wherein the dmixloudoffst indicates whether there is adifference between the expected loudness of a 2-channel downmix outputfrom an AC-3 or Enhanced AC-3 decoder, and an actual measured loudness,with all gain adjustments due to dialogue normalization, dynamic rangecompression and/or fixed attenuation to protect against downmix overloadhaving been applied prior to measurement.
 8. The audio processingapparatus of claim 5, wherein a program loudness data conveys data aboutprogram loudness, including measured loudness values for the program,and indication as to whether the loudness of the program has beencorrected prior to AC-3 or Enhanced AC-3 encoding.
 9. A non-transitorycomputer-readable medium comprising machine executable instructionswhich, when executed, cause the machine to perform steps of: receivingan encoded audio bitstream, the encoded audio bitstream including audiodata and metadata audio signal including one or more sets of downmixloudness parameters; determining if a first downmix loudness parameterindicates whether downmix loudness offset data exists, and if thedownmix loudness offset data does exist, adjusting a difference betweenan expected loudness of a downmix audio signal and a measured loudnessof the downmix audio signal based upon a second downmix loudnessparameter; and wherein the second downmix loudness parameter indicateswhether there is a difference between the expected loudness of a2-channel downmix output and an actual measured loudness.
 10. Thecomputer program product of claim 9, wherein, the first downmix loudnessparameter is a one-bit data field denoted as “dmixioudoffste” and thesecond downmix loudness parameter is a five-bit field denoted as“dmixioudoffst.”
 11. The computer program product of claim 10, whereinthe dmixloudoffst indicates whether there is a difference between theexpected loudness of the 2-channel downmix output from an AC-3 orEnhanced AC-3 decoder, and the actual measured loudness, with all gainadjustments due to dialogue normalization, dynamic range compressionand/or fixed attenuation to protect against downmix overload having beenapplied prior to measurement.
 12. The computer program product of claim11, wherein a program loudness data conveys data about program loudness,including measured loudness values for the program, and indication as towhether the loudness of the program has been corrected prior to AC-3 orEnhanced AC-3 encoding.